Re: [asterisk-users] Asterisk to record CDR in DB Oracle
Hi Everton, Everton Goularth wrote: I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome There is no native Oracle driver available to my knowledge, but if you can install an ODBC driver for Oracle, Asterisk will happily use that. Best regards, Florian ___ --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] jittery audio in voiceprompts
Hi Murf, Jason, Steve Murphy wrote: I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. What do you have installed, that will provide the 1Khz timing interrupts you will need to function properly? Actually, I doubt the timing source will be required if you only use playback or background commands with the supplied gsm prompts. We run lots of machines without it. Timing sources are used for some cases of musiconhold, meetme and the likes, but not for regular stuff. Jason, if you do a 'vmstat 1' on the unix prompt when a call is run, does it ever hit an idle count of 0 somewhere ? If so, you have performance issues, if not, you'd probably look toward the network, or perhaps a silly Voice Activation setting in your phone. If possible, you could also try and look at a tcpdump capture of your traffic using wireshark to see if there is specific jitter or packetloss in the audiostream as it leaves the server. Best regards, Florian ___ --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] Transfer on RTP timeout?
Hi, Ray Jackson wrote: transfer to that number. That way the call can stay up rather than the user having to redial. Is there a way of transferring back to the * dialplan on RTP timeout to perform some additional steps (instead of just hanging up?) Nokia seems to have done something like this in their E-series (E60 etc) with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ? Florian ___ --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] Dual Ringing Tones
Hi guys, Leo Ann Boon wrote: I have a couple of interconnected asterisk boxes connected to several providers. With one provider in particular (ATP in Australia) there are two ringing tones heard on outbound calls. It is not the end of the earth - I am not reselling our services yet - but it is strange being that none of the other providers we are connected to exhibit that behavior. I think your provider is providing early media. Check your sip messages, look for 183 with SDP in the response from the provider. Correct, early media is offered when this occurs. Solutions: 1) Add the 'r' parameter to dial, which causes asterisk to fake the ringing signal. You will lose early media in the process 2) Modify the Dial command like was done in BRIstuff so that the ringing is faked by Asterisk _only_ if the remote party indicated ringing too. Or is there a more correct approach ? Florian ___ --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] What web interfaces are available today for debian based Asterisk installation?
Lee wrote: Maxim Veksler wrote: I am aware of both of these tools, I don't like them! They make absolute changes in your /etc/asterisk/* files, they assume that they are the only thing you will be using for managing your asterisk pbx and they are both totally unfriendly to 3rd party changes. Yup, which is precisely why the webtools we built (see post from Michiel, thanks!) will only write into separate files that can be #included. Florian ___ --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] Siemens Gigaset C450 IP vs S450 IP
Hi Eugen, Eugen Leitl wrote: I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? Unfortunately I haven't yet had one in my hands, but from the feature-list it seems a bit more value for money compared to the C450. Especially being able to handle 4 SIP accounts/lines at one provider, being able to add more handsets to the basestation etc. would be of value for SOHO use. I did notice the C450 is unable to use the flash-key for call transfer functionality with SIP accounts, which is a bit of a shame. I'm not sure if that will be supported with the S450. If anyone can shed a light on that I'd be interested as well :-) Florian ___ --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] Mediatrix 1204
Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --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] Asterisk in Xen 3.0
Hi, Tomer Horn wrote: Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? We do this a lot, although I believe our engineers are still using Xen2 for systems with BRI/PRI adapters. Xen3 is fine if there is only software involved. Florian ___ --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] Samsung Prostar DCS
Curt Shaffer wrote: I walked into a new potential * install yesterday. They are running a Samsung Prostar DCS. Does anyone have any experience with these out there that you could relay some things to look out for when integrating this until the migration is complete? Or what would be the best way to integrate it while migrating. We coupled an Asterisk box to a Samsung DCS a while back. We did it based on ISDN2 BRI lines, where Asterisk was the NT side. We did notice the DCS had some problems accepting different timing sources on different TE busses, causing some strange effects with callers in meetme-conferences. It worked fine for some time, but needed recycling every once in a while, so I would not recommend this approach unless you also have a good DCS technician with you :) Eventually we phased out the DCS and implemented IP-phones all over the office. If you need a solid approach, perhaps you can place the *-machine in the path of the ISDN lines: DCS - * - PSTN This would avoid the timing issues we have seen. Best regards, Florian ___ --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] Re: Load balancing of IAX2
Hi, Kamran Ahmad wrote: I have a question in this case when call is transfered from loadbalancing-server to server01 or server02 what will be media Path? media will be routed through loadbalancing-server or it will not use loadbalancing-server anymore EndPoint1--loadbalancing-server--server01/02--EndPoint2 OR EndPoint1--server01/02EndPoint2 That depends on the exact configuration. If the loadbalancer is an IAX machine it can remove itself from the IAX path if that is allowed in iax.conf (check parameter 'notransfer'). Florian ___ --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] Load balancing of IAX2
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten = _X.,7,Dial(IAX2/server01/${EXTEN})
exten = _X.,8,Congestion
exten = i,1,Congestion
Florian
___
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Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Michiel van Baak wrote: If you buy a model without the spare in it's name, you have the license to use them right ? To use them with a CCM or CCME, yes :-) How about secondhand phones you get from ebay ? Is my cisco smartnet account enough to run the phone legally ? It's not a spare model (at least that was not in the deal description) My understanding is, if you have any license at all, Cisco will probably not bother you. But it is most definitely not the way they intended :-) Florian ___ --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] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Cory Andrews wrote: In my interpretation of the oft confusing Cisco licensing structure for phones, the license was originally created to function much like a COA with a piece of Microsoft software. When adding a client phone to a CallManager or CallManager Express network, the user is required to have a license for the client phone. Cisco phones are sold pre-bundled with a license in their CH1 form. If there is a CH1 or CCME attached to the part number, it is a licensed bundle. Cisco also offers spare versions of their phones, which do not have the CH1 or CCME annotation to the part number. These are unlicensed phones. Cisco phones do not currently ship with SIP firmware loaded. In order to register your phone with Cisco and obtain a login for their TAC and access firmware downloads, you must have a licensed phone. Spare versions do not allow you registration with Cisco for access to firmware. Point is, you do not really need a CH1 or CCME license, you are free to combine the Spare phone with a separate SIP license - the price is identical. It is NOT OK however to use a Spare phone without any license, as far as I am aware. Florian ___ --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] Do you need a licence to connect a Cisco hardphone to Asterisk ?
Olivier wrote: Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=) along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to connect it to a SIP enabled Asterisk server ? Yes, as far as our sales rep can tell us. Florian ___ --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] isdn-data over iax
Hi, [EMAIL PROTECTED] wrote: is the following zaptel.conf configuration correct for TDMoE used for pri-cpe signalling - is this possible at all ? I couldn't find an example... Any kind of Zaptel signalling should be fine. Check out http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE Best regards, Florian ___ --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] Echo Cancellation
Hi, Douglas Garstang wrote: If you install a Digium card in an Asterisk system, and install zaptel drivers, do this give any benefit of echo cancellation? Our PSTN gateway is a separate Audiocodes box, so the zaptel card wouldn't actually be connected to anything. I'm wondering though doing this would help, in general, with echo cancellation. No, if your telco lines are not connected to the zaptel card, the zaptel driver and echocancellers will not help you one bit. Best regards, Florian ___ --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] isdn-data over iax
[EMAIL PROTECTED] wrote: is it possible to route an ISDN-Data channel over an iax-connection ? the setup is pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk Server2 (E1)-connecting to an external isdn-dialin router via the iax-line the call is transfered as speech which is not accepted at the remote end IAX is not suited for this. Maybe TDMoE is an option for you ? Florian ___ --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] MOS Scores and LCR
Hi, trixter aka Bret McDanel wrote: MOS (Mean Opinion Score) is generally a bunch of people sitting there listening to audio and rating it 1-5 (there is a newer method that is twice as good becuase it goes 1-10, basically all values are double). Its their opinion. This generally cant be dont automagically and still be MOS. You can try to track frame drops and other things on your end to rate call quality and try to come up with something, but that technically isnt MOS. AFAIK asterisk doesnt keep statistics of jitter, frame drops or anything else, that might be a good project for someone to take on, especially if you have multiple providers so you can rate quality in a more meaningful way. The human ear really isnt the best tool for much of this. There are ways to guesstimate MOS scores on a call by continuously getting some decent statistics from the jitterbuffer. We've had an intern do some work on this using IAXclient. http://www.speakup.nl/en/opensource/jitterbuffer/ Florian ___ --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] MOS Scores and LCR
Hi, trixter aka Bret McDanel wrote: yes and I suggested that however, MOS is an opinion, so its totally subjective and not based on anything 'real'. That was kinda my point earlier. Personally I think that its better to isolate the network/cpu issues and correct them to get what a given implementation of a codec is supposed to be rated at (ideally the two would be intertwined). Technically you are right, but its difficult to communicate that in the market. Using 'accepted' methods like MOS (or variations thereof) makes discussions with other parties a lot easier. The work that you have done so far is a great step towards a product that many people might find useful. In a nutshell the concept I am thinking about is a tool that you drop onto your network and it will monitor the data (presumably not just iax but sip, h.323, whatever) and generate live stats of the call and possibly even have an alarm system that would send off a page or something if conditions get too far from 'normal'. Yes, that would be excellent indeed. Problem is that the location of measurement will influence the scoring :) If you have good ideas towards this we'd be very interested in participating. Florian ___ --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] Hard drive write cache
Hi, shadowym wrote: I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Things to consider: - Use compactflash to boot and run asterisk, add disk only for voicemail - Run the entire setup from a ram disk, make commit/rollback facilities to write to disk - Extra servers are cheap - you could use LinuxHA to failover the server. Florian ___ --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] Block access to [EMAIL PROTECTED]
Pietro U wrote: i have a problem, if i dial [EMAIL PROTECTED] i can call my doamin users without any registration in the asterisk. how to block this? Point your default value in sip.conf to an empty context. Florian ___ --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] Nokia E60 , experience as SIP client
John Joseph wrote: Hi I want to check out from the members , about their experience with Nokia E60 phone as SIP client , I was able to register the phone , but my voice gets broken during the calls . My other Wi-Fi VoIP SIP phone are working fine I also like to check out is there any other mobile manufacture who have SIP supported porducts like Nokia e-60 We use them with Alaw/Ulaw and it works pretty well. I do think there are some bugs in the firmware, SIP accounts do not get reregistered automatically if other applications used the WLAN network, or when roaming between different WLAN networks. I'm also not entirely happy about the battery time when using WLAN :-) Great phone, though! Florian ___ --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] VLAN info
Hi, Citeren Alan Neville [EMAIL PROTECTED]: I'm looking for any information on setting up VLANs to seperate the telephony network from the ordinary network. I have google'd around but haven't found a lot of information in the best way to go about this. Has anyone managed to do this successfully in conjunction with asterisks? If so, could they provide an overview of what they did and how they find it? Did the performance improve after the VLANs were setup? Using VLAN's (and more importantly, VLAN priority settings) will most definitely be able to improve VoIP quality. In Linux, a VLAN will be another logical ethernet interface, and thus, to the configuration of Asterisk it makes no difference. Take a look at: http://www.linuxjournal.com/article/7268 -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ --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] Netherlands zaptel.conf
Michiel van Baak wrote: If you load the wcfxs module and everything works (cept for the asterisk answering the phoneline) all is correct. wcfxs is for connecting an analog phone, not a PSTN connection. I think you have the wrong module on you wildcard to interface with the PSTN net. Sorry. Whoa, good call! I totally ignored that options. Pieter, what color is the module ? S110M = Green (for a phone device) X100M = Red (for a phone line) Florian ___ --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] Netherlands zaptel.conf
Hi Pieter, Pieter Claassen wrote: Well, I tried to plug my KPN phone line into it as well with the same result. The PC refuses to answer using the fxsks protocol. I don't think these phone lines are IP carriers and suspect that UPC might turn the voice stream into something else in their modem. The phone however is a standard analogue device and I suspect you can stick anything you buy over the counter in there. Correct. AFAIK, the most notable difference is UPC offers an FSK based CallerID instead of the ETSI DTMF mode used by KPN. Even though UPC will use some form of VoB transport within their network, the connection you as a customer are facing is most definitely a regular analog line. I haven't personally played with the TDM series interface cards, but here are a few thoughts: Config included below. The question is how to start figuring out what is going on since I don't see any messages in /var/log/asterisk/* or syslog that indicates there is a problem? lsmod includes zaptel225284 1 wcfxs Hmm, shouldn't you be loading wcfxo ? The configs look sane enough, most things matching with an old X100P setup we have laying around. Most of my work is on BRI/PRI systems though :-) Florian ___ --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] Dialling a DUNDi Route
Hi, Douglas Garstang wrote: We are using a backend MySQL database for call flow, not user agent registration info. Just how, exactly, is a backend database going to replicate registration data between Asterisk servers? Realtime has been documented NOT to work with multiple Asterisk systems. If you like I can dig up the list messages from Kevin Fleming on this subject. Realtime also has way too many limitations. You're thinking inside the box. I'm not saying Kevin is wrong. You can probably design a database that uses a per-asterisk set of tables and uses triggers or a stand alone daemon to manually replicate the data between machines. If realtime doesn't fit your need, consider automatically generating extensions.conf etc. from databases using scripts and templates. F. ___ --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] Dialling a DUNDi Route
Douglas Garstang wrote: No... do you have an example of what that looks like? I get more matches on google for 'the early history of hungarian cabinet making' than I do for DUNDi examples. [dundi] type=user dbsecret=dundi/secret context=dundi-e164-local Best regards, Florian ___ --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] Dialling a DUNDi Route
Douglas Garstang wrote: We're doing all of our call routing from a database accessed from AGI. When we trunk calls from one asterisk system over to another via IAX to terminate the call, the dialling parameters are defined by what's in the dial command on the second system, not the first. This is a big problem. :( Eh, ok, I have a very faint idea of what you are saying. But what are you trying to achieve ? F. ___ --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] Dialling a DUNDi Route
Douglas Garstang wrote: What am I trying to achieve? Uhm... a carrier grade, highly redundant (ie multiple servers), VOIP solution with advanced business(not residential) features such as findme/followme, incoming and outgoing blacklisting/whitelisting(user/org/company level), user/prefix defined pic codes and rate centers, intra company 4 digit extension dialling, feature codes, user defined internal, external, override caller id and on and on - all provisionable and maintainable via a web interface (don't forgot the multiple servers!)! does that answer your question? Yaddayaddah. Don't go at this lighthearted. You can use DUNDi for call distribution between asterisk nodes and automatic discovery. However, depending on how big your site(s) will be, it may be worthwhile to take a look at database integration (i.e. the realtime API in asterisk). It will in most cases give you a finer level of granularity than DUNDi will. When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the ring time and ring options of the original SIP call between servers. Correct, same applies for using 'switch' in your dialplan. Once the call is gone, it's gone. DUNDi was not designed for that type of applications. Maybe you are better served with a good dynamic database on multiple servers :-) My EUR 0,02 F. ___ --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] Junghanns, Germany ISDN settings
Hi Chris, Chris Earle (CBL) wrote: Thanks for the info, I am confused still ;-) It sounds like I need NT mode -- there are NTBA boxes involved at my location... No, thats the point: If your telco delivers NT boxes, your equipment must use TE mode. It's always a pair: One side does NT mode, the other TE. Termination of S/T Interface ?? Usually you don't need to bother with that, the factory setting is fine. Power Feeding? Only needed in NT mode in combination with ISDN phones that require power feeding. Doesn't seem nessecary in your case. Best regards, Florian ___ --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] Junghanns, Germany ISDN settings
Hi Chris, Chris Earle (CBL) wrote: I've got a Junghanns QuadBRI card which I'm going to install on a system in Germany Anyone give me some tips on the Jumper settings? I'm guessing it's going to be NT mode with p2p? I haven't used ISDN before. I'm going to also put a Digium TDM400P card in there to plug the analog phones into. I'm just worried about the jumpers and modes. It really depends what you will be hooking up to the asterisk box. If you are connecting to a telco's S0 bus you want the card to be in TE mode (Terminal Equipment). If you are using multiple ISDN lines that are coupled together as one bundle (ask the telco) you will probably neet to configure it as p2p. If all lines are singular, use p2mp. If you will be connecting to a PBX, everything is dependant on how that PBX is configured. Best regards, Florian ___ --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] Set CallerIDNum on a PRI
Hi, Mimmus wrote: I have a PRI line with DID (from 100 to 499) in Italy. Now I'm seeing all calls from same DID 'main' number. Can I set outgoing CallerIdNum to the right extension? Yes, assuming your telco allows you to. Be sure to figure out what number format is required in your case. Your telco can tell you. (Often this is the full DID without a leading 0) Best regards, Florian ___ --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] Codec negotiation
Hi, Ronald Voermans wrote: I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming phonenumbers. One phonenumber is for voice-calls, the other one for receiving faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the fax-number by G.711. Can I make a codec-negotation based on the called number? Nope, but maybe you could separate the traffic in to different SIP peers. If you need more info on this, i can send it to you. If you want we could figure something out. Just curious: Which PSTN provider are you using ? Florian ___ --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] Codec negotiation
Hi Ronald, Ronald Voermans wrote: What exactly do you mean by seperating traffic in to differt SIP peers? The situation is as follows: I have OpenSer connected to our SIP provider/PSTN Provider (the answer to your question: Enertel). Ah 'kay. Asterisk registers to OpenSer, which then forwards the call to PSTN. Asterisk registers two numbers at OpenSer; one phonenumber and one faxnumber. I also made two entries in sip.conf. However, the host=... Is the same for both numbers. So incoming calls are always matched to one (1) peer/entry in sip.conf. Hence the problem with negotiating the right codec (g.729 for voice, g.711 for fax). Hrm, yes for inbound the problem is with the host=.. matching. Maybe Olle has a good suggestion on this :-P. However, if you control the OpenSer yourself you could easily bind another IP, or perhaps use OpenSer rules to do the trick ? Asterisk SIP stack doesn't seem suited for this type of traffic separation I guess... Florian ___ --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] How to handle provider UNREACHABLE in the dialplan?
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is exactly the point what I am looking for. How can I use the next
peer in the dial logic? I was trying DIALSTATUS, ... but I could not
make it.
Should be easy; we use:
[macro-safedial]
;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4})
exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
exten = s-NOANSWER,2,Hangup
exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1})
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1})
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
Florian
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Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
Hi Ronald, Ronald Wiplinger wrote: voipbuster/ 194.221.62.201 5060 UNREACHABLE voipstunt/x 194.120.0.200 5060 a reload shows than: voipbuster/ 80.239.235.200 5060 UNREACHABLE voipstunt/x 194.120.0.200 5060 UNREACHABLE Seems like voipbuster is doing round-robin DNS for redundancy. Bad choice with asterisk, since asterisk only looks up DNS on startup or reloads. You could read out all the entries in the DNS zone and create your own list of entries in /etc/hosts, and then create multiple asterisk peers: voipbuster1, voipbuster2, etc... Then you can use regular dialplan logic to cycle through all of them. Florian ___ --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] Asterisk + XEN does it make sense?
Jean-Michel Hiver wrote: Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? We run a number of systems with Xen, its great once you figured out the nags of it :) Remember, to do anything with hardware you will still need Xen 2, not Xen 3. Florian ___ --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] simulating a few thousand SIP clients?
Roy, Wai Wu wrote: sure, but I need to simulate the SIP REGISTER and OPTION traffic sent by ATAs as well. What is the current registration time you accept on the servers ? 3600 ?? One thing you can do to try this is set a number of devices to a much shorter registration period. This will effectively deliver just as many REGISTER commands so it can be used for a reasonable test. We've used 10 phones at a registration time of 1 second to 'emulate' 1200 phones at a registration time of 120 seconds. This will ofcourse not emulate the call volume, only the REGISTRERs (and perhaps OPTIONs). Florian ___ --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] DEFAULT_USERAGENT
Hi, Thczv F. Thczv wrote: Would there be any other nasty consequences of making that change? More importantly (perhaps), is there any way to make the change in [EMAIL PROTECTED] without doing a recompile (and potentially screwing up my system beyond my ability to repair it)? We modified this on a few of our servers, without any noted ill-effect. It's even user configurable in sip.conf: useragent=My First SIP UA Best regards, Florian ___ --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] DEFAULT_USERAGENT
Hi, Thczv F. Thczv wrote: Would there be any other nasty consequences of making that change? More importantly (perhaps), is there any way to make the change in [EMAIL PROTECTED] without doing a recompile (and potentially screwing up my system beyond my ability to repair it)? We modified this on a few of our servers, without any noted ill-effect. It's even user configurable in sip.conf: useragent=My First SIP UA Best regards, Florian ___ --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] E1 Echo
Rich Adamson wrote: We have found that a relatively innocent change by the local incumbent operator has forced us to modify our pstn gateways to change from 128 taps to 256 taps. What type of a change did they make? Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. ___ --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] E1 Echo
Andrew Kohlsmith wrote: On Friday 16 December 2005 08:12, Florian Overkamp wrote: Although it's a bit unclear how things evolved exactly (since no-one ever tells us), a number of interconnection points throughout the country were consolidated, significantly increasing the chance that delay exceeded 128 taps. I need to do some investigation of bringing the tap count WELL above that... I'd like to see what kind of performance we can get with 128 MILLISECOND tail... 128 taps is only 16ms... and 16ms of echo cancel is damn near useless, as it's fast enough that you'd likely not even hear the echo as anything more than a sidetone anyway. I imagine it's deathly hard on the CPU though. :-) Actually, the problem is different. If you receive an echo on the PSTN gateway that has a 16ms echo, the problem would not be noticeable there, but if you then add a VoIP connection the delay added would make the echo audible. Florian ___ --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] E1 Echo
Rich Adamson wrote: Strange... I would never had expected consolidation to have that kind of impact. It almost sounds like they have something in the E1 data stream that buffers (and delays) content, maybe decoding and re-encoding in some fashion. Well, the problem is the difference between keeping under 16ms and sliding _just_ over limit to 18ms would make the effect audible almost immediately. We used the sangoma echospike tools to measure the delay and adjusted our taps accordingly. Florian ___ --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] E1 Echo
Hi Rich, Rich Adamson wrote: Sangoma echospike tools? Please elaborate! See sangoma's -users posting from Dec 13th, which I quote: I just wanted to let you know that we do provide a tool to debug echo. We send a unit impulse and record the Finite Impulse Response (FIR) so it can be plotted and analyzed. The code that does this is the release at ftp.sangoma.com/linux/custom/2.3.4. Instructions on using it are in the wiki in http://sangoma.editme.com/wanpipe-linux-asterisk-debugging. Although the code is wanpipe, all the interaction is at the zaptel level, so I am pretty sure it will work on Digium or other cards as well. Just being able to see what the echo looks like on a troublesome line gives quite a lot of info. You can see if the echo is delayed, or markedly non-linear. I haven't tried it as yet but plan to do so. Correct, this is what we used. Florian ___ --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] E1 Echo
Hi, Rich Adamson wrote: I am beginning to wonder whether what echo IS heard is being caused by packetisation delays in the network - The default tap length is 128, or I believe 16ms. If something in the PSTN causes a delay more than that length (no idea what might cause that) then echo would still be heard. We have found that a relatively innocent change by the local incumbent operator has forced us to modify our pstn gateways to change from 128 taps to 256 taps. Since th Does anyone have any experience in this area? Any ideas? How heavy handed would it be to increase the tap length to 256? I have not seen anyone suggest that this might be a good idea. There have been a few issues especially related to the echotraining section (which can go boo-boo on E1 lines because the audio path is not always entirely complete when zaptel expects it to). If you make sure you are on recent zaptel EC standards you can up to 256 taps. There will be a minor residue that needs work, but it will allow a lot of room to decrease the loss-plan you may be using now. Florian. ___ --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] callerid international-format
Florian Meister wrote: Hi, Is it possible to send international format (+435572999888) with asterisk. I have the following problem: When I set the calleridnum to the format above, the telephone (grandstream ata with a siemens gigaset) does not display the +. So I send it now with 00 instead of the + for the international prefix, but it would be nice if it would possible to make the +-thing work. You could try messing with the type of callerid the ATA is sending. In DTMF you cannot send a '+' symbol, but maybe in Bellcore it can work ? (For the record: I doubt that this is possible, but feel free to try) Florian ___ --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
[Asterisk-Users] app_queue on 1.2 ?
Hi We're trying to migrate our platform from 1.0 to 1.2 and we're seeing some oddness in app_queue. We use local_channels a lot for things like persistent agents, call-forwarding on agents and such. Now on our 1.2 server we notice that the queue is listing all members as 'Invalid' (thus any caller will automatically fall out of the queue, no phones will ring) until we issue a reload manually: After startup: omega*CLI show queue queue1 queue1 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet Local/[EMAIL PROTECTED]/n (Invalid) has taken no calls yet No Callers omega*CLI After we issue a reload: omega*CLI show queue queue1 queue1 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet Local/[EMAIL PROTECTED]/n (Unknown) has taken no calls yet No Callers omega*CLI I cannot find anything that explains this in the changelogs. I'd appreciate some comments on why this is, and how it can be fixed (other than completely redesigning the platform, which I do intend to do, but not just now yet) Florian ___ --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] app_queue on 1.2 ?
Hi Philipp. Philipp von Klitzing wrote: Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Thanks for the suggestion, but I don't think so - this is fresh a 1.2.1 svn checkout. I will see if it gets cleared without the /n Florian ___ --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] app_queue on 1.2 ?
Philipp von Klitzing wrote: Hi Florian, have you check that this is not connected to bug 5810? Just a guess. Checked and verified, the patch from 5810 is properly applied in my 1.2.1 checkout and the issue remains with and without the /n. Any hints ? Thanks, Florian ___ --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] Re: Can anyone explain reason for this echo
Hi Eric, Eric Bishop wrote: I purchased the following item: http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html As you can see not a very highly spec'd product but does the job well. Can you indicate price range for this unit ? Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service
Hi Frederic, Not to start some flame war here, but I've always known the Junghanns people to be quite cooperative, although it is a shame that they don't have two Klaus'es around there, since one is just simply too busy :) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi, FaberK wrote: during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 So, is FLEX available on your system ? (I don't know CentOS) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi, FaberK wrote: Hi Florian, yes, I have Flex available: whereis flex flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz Hmm, nope sorry :P. You can try to mail or call Sangoma, their support is pretty good from what I've seen so far. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bizarre Echo Problem
Hi Mark, Citeren Mark Edwards [EMAIL PROTECTED]: to add some fuel to the fire, I was monitoring one of the agents last night. He made a call to a target and then had to call them straight back to confirm some information. The first call was as echoey as the inside of a cathedral. The second (next) call was as clear as a bell. Although I did not follow your thread from the start, I can confirm there are significant differences in zap-channel inbound versus outbound echocancellation, i.e. the direction of initiation of the call is important. As far as our tracing went, the echo-cancellation itself was properly activated on the channel, but the exact moment of canceller-training may not be perfect. Think of when training can occur _before_ the entire call-path is up and active. This would mean the training is infact useless and may even be increasing problems. We have been managing the issue via implementation of a small loss-plan and some mods to chan_zap, although I need to see with our developers on how that was done precisely. Best regards, Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
Hi, Citeren tim panton [EMAIL PROTECTED]: PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! I've mirrored it on our website at http://www.westhawk.co.uk/resources/AsteriskTFOT.zip And another mirror: http://www.speakup.nl/en/opensource/asterisktfot/ -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
snacktime wrote: permit to be used for their contributions.. They won't be happy unless everyone else does things their way. They wouldn't be happy if asterisk was BSD or MIT licensed either. No that's not true. I myself would be perfectly happy with an MPL. However, because Asterisk is available under a GPL formed license, any fork will need to be GPL too, until such a time that any and all GPL code has been replaced by something the prospective owners are willing to relicense under something else. FLorian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
Hi, Michiel van Baak wrote: What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive I got mine from www.centralpoint.nl As far as I know they only deliver the phones with SCCP image. But as you can read in my previous mail this is no problem, simply install chan_sccp. If you want the phones to run SIP, you have to buy a license for the SIP image. Centralpoint has them too. My company is a cisco supplier too, maybe we can arrange some pricing strategies together. However, Cisco remains an expensive phone. Be aware, you cannot really compare delivery from any dutch supplier to what you find on Ebay. We only deal in new stock, nothing refurb, and yes, they are expensive. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] slight echo via sip provider
Hi, Damon Estep wrote: Here is the setup; analog phone Linksys ata asterisk sip provider sonus GSX 9000 PSTN called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight echo of their voice. All of the Zapata.conf echotraining, echocancel, etc do not seem to apply here as there is no zap channel involved in the call. Correct. I assume that since the echo is toward the called party who is on the other side of the provider sonus softswitch and somewhere on the PSTN, that the echo is really coming from the providers media gateway/softswitch. This is possible, but not really likely. Most decent service providers use digital equipment and would (should) not introduct additional echo on their end. However, it is very well possible that your Linksys ATA and the connected analog phone are causing the echo. I'm not sure about the capabilities of the Linksys, but with Sipura's you can modify the line impedance settings to best match your equipment. Look for the Regional Tab at the top. There is a setting called FXS Port Impedance. Try various options in there - they should match your phone. Best regards, Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX and AGI
Hi, Daniel Grad wrote: I am writing a script (php script that runs via fastAGI) that takes incoming calls and processes them in various ways depending on settings from a database. At some point, I need the script to receive an incoming fax. But the problem is that if I run NVFaxDetect from the script, then asterisk crashes. If I run rxFax without NVFaxDetect, then I get errors when sending the fax. What can I do? How can I receive a fax with a PHP fastAGI script? You could use a goto to exit the AGI script first and jump to a fax reception context/extension. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DECT gateways
Michiel van Baak wrote: Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them with asterisk. I been looking around on the internet and found the Kirk gear. Anyone has any experience with them ? The website states they are recognized as Cisco 7970 in CCM. Does chan-sccp handel those Kirk emulated devices ? Hi Michiel, I have a Kirk set which should be able to do H323, but I haven't had time yet to try it. They have SCCP and H323 types, and ofcourse there are sets which can be connected via an E1 link. If you have time I'm sure we can figure it out :-) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DECT gateways
Yoann Le Bihan wrote: 2005/8/17, Michiel van Baak [EMAIL PROTECTED]: Is there any other solution like this out there that works with asterisk ? Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such expensive compared with Cisco ones...) ? Because if you have a network of DECT (maybe even GAP) repeaters, why should you invest in a new WIFI network ? Besides that, in most WIFI basestations and handsets things like handover and roaming are not yet good enough to be accepted by demanding end-users. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation again ...
Rich Adamson wrote: I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Keep reading. Echo _can_ occur whenever a two-wire circuit is converted to a four-wire circuit (eg, hybrid involved). There is no way for you to identify where (in a pstn call) that might occur even with a PRI. You could have a PRI (four-wire) leaving your facility, but the telco (or another pstn customer) may have a hybrid involved in that end-to-end call. Very nice summary post. One thing to remember: If your side of the call is hearing an echo, it is being caused somewhere on the other side. Also, there are two kinds of echo. Hybrid echo, which has now been discussed, and accoustic echo. Any echo can be batteld best at the place where it is caused, and this is especially true for accoustic echo. With accoustic echo, see if the source device (handset, speakerphone, ...) can be tuned properly (echocancellation settings, echosuppression or maybe just lowering the volume a bit). Also note: An echo becomes more and more problematic as the round-trip delay time increases. Best regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Hi, Ronald Voermans wrote: If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? This is very much dependant on how your dialplan works. We use normalisation for each account so the system doesn't have to worry about many different dialling formats (i.e. with or without areacode, and such). You can use a similar strategy for all your internal numbers as well. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_Queue strategy=ringallfree (feature request, possible bounty)
Kevin P. Fleming wrote: Kristian Kielhofner wrote: Not having looked at the code (like I could make much sense out of it anyways), how hard would it be to add something like strategy=ringallfree, where only members of the queue not already on a call from that queue will receive incoming calls? We have been suggesting that people implement this sort of thing by using Local channels and the dialplan, rather than trying to force more complicated logic into app_queue. By using the dialplan, you can use any method you wish to decide that the agent is 'busy'... look in a database, run an AGI, etc. Hi, this is a viable option, I have actually defined persistent agents in older asterisk versions with that strategy. It does seem to have considerable effects on the logging of the calls though, so if queue analysis is important you may get more workload than you bargained for. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like tosee ina GUI?)
Hi, Sherwood McGowan wrote: I personally prefer MySQL-MAX. I curently run *RT in a large production environment comprised of more than 1K users, with MySQL-MAX as my backend. Also, it's a point of I've spent so much time working with MySQL that I don't want to have to jump systems. It's fit the needs of the VOIP provider I work for and causes no problems that I see, so if it ain't broke, don't fix it is the rule here ;) Many people like many DB's for many different reasons. I for one would appreciate any design where the database functionality either: - is using an abstraction layer so many DB's can be used, or: - is designed so all direct DB interaction is in one centralised place so rewriting for a different DB becomes a manageable task. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vizufon Video Phone
Hi, -Original Message- So I won one of these on ebay, in the auction it says it has the RJ45 ports on it but it doesn't :( If I were to get an analog adapter would I be able to use the video portion of this or am I SOL? The auction requires me to pay for shipping back, so I end up losing money unless I sell it on my own. Or has anyone hacked these to work on enet? The Vizufon is a 21st century video-over-analog desk telephone that offers the best performance and the lowest cost for a video telephone anywhere in Detailed entry And later on: Network - PSTN (Public Switched Telephone Network) RJ11, Transmission speed(max) : 33.6Kbps(V.34 bis), Communication mode: Synchronous access This leads me to believe this phone is actually a completely analogue device that performs modem signalling to transfer video. At 33.6Kbps you can forget about using that over a VoIP ATA. So if you bought this under the impression it was an Ethernet SIP device of some sort, you may have been misinformed. Sorry... Source: http://www.veilingtips.nl/linkupgold/links_detail/link_detail_1331.html (BTW, I found that via Google: http://www.google.nl/search?hl=nlq=vizufon+videometa=) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] swissvoice
Hi, -Original Message- I have swissvoice phones and when i use one, a have in asterisk lines like: Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp -13691.-232125 the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and asterisk version is the cvs of 18 july 2005 (today). Swissvoice phones tend to have a few interesting side effects in their rtp timestamping, we have filed some issues on that. However, it would be fun to hear what the actual problem is you are experiencing :-) Best regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN to SIP gateway
Hi, -Original Message- So far I've gotten Asterisk to say: -- Extension 'XX' in context 'pstn' from '' does not exist. Rejecting call on channel 0/23, span 1 (where XX is the phone number I dialed) So, that's a start, I guess ;) extensions.conf contains: [pstntosip] exten = _X,1,Dial(SIP/[EMAIL PROTECTED]) I'm sure I'm missing a vital config option elsewhere... Try: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) The '.' is a wildcard match of unknown length. With your pattern you only accept extensions of 1 digit long. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video phone settings???
Hi, -Original Message- disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Have you tried permutations of this ? I have had working setups with everything except h263p. My experience with leadtek phones is they tend to crash when they are talking to any phone model that is not exactly the same (i.e. bugs in decoding perhaps ?) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] presence and IM again, want to develop a workinghack
Hi, -Original Message- I personally don't think it's a good idea to implement it in chan_sip. One reason for this is that user1 wants msn, user2 wants jabber, user3 wants icq, user4 wants gadugadu etc etc. Are you gonna implement all this ? That is, if you mean Instant Messaging in SIP ;) Forgive me if I'm wrong... You actually need a little of both. For PC interfaces, a universal messaging would be nice. There are SIP devices which support SIP messaging though, and they will not be able to deal with much else. Some form of integration in chan_sip will make sense... Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wi-fi phone advice
Hi, -Original Message- this morning a got a message, that you can by a F1000 from UTStarcom at sipgate.de (Online-shop) for EUR 169,- That's not bad at all. Has anyone used these with asterisk yet ? I have a few WIFI devices, but they tend to loose registration every once in a while. Could be my basestations too, tho. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicexml
Hi, -Original Message- Does asterisk have a fully working (or anything in active development) voicexml parser? I have looked and if there is anything google isnt being friendly to it. I was considering writing one if nothing existed, however I dont want to reinvent the wheel. Phonologies (http://www.phonologies.com) says they have something in that direction, but its commercial. I don't think any open solutions are available, at least not in any state of maturity. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID problem..
Hi,
-Original Message-
what i mean is, i make a call from another did number
but people receive the pilot number.
i don't know how to do :(
i try this but nothing happen.
exten = _01,1,SetCIDNum(0${CALLERIDNUM})
exten = _01,2,Dial(${TRUNK}/${EXTEN})
please help me..
It is very much dependant on what type of line (analog, T1, E1) you have and
what your telco allows you to do, so ask your telco.
In many of our setups, it is required that callerid's are sent in a specific
format, the national number without leading 0 to be exact. Similar
requirements may apply to your setup.
Talk to your telco.
Florian
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RE: [Asterisk-Users] GSM Hunting
Hi, -Original Message- Need to implement hunting (create a hunt group so my subscribers can have a single GSM number for access to me)of GSM SIMs on a GSM bank independent of the Telco for the SIMs. Anyone got an EXACT idea how to do this? If you want 1 GSM number that can access many GSM SIM's you need assistance of the GSM telco. Alternatively you could enable call forward on busy for all the SIM's, so they daisy chain through there. Might end up being a very expensive solution though... Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel HEAD with * Stable?
Hi, -Original Message- Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? Err, why specifically would you want that ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel HEAD with * Stable?
Hi, -Original Message- -Original Message- Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? Err, why specifically would you want that ? In our case, the CVS drivers (At the time that I did it) showed enhanced information coming across our Definity PBX, before we wern't getting CID info and we are now. Ah, okay. Thanks for that addendum. It's good to know this is possible, it may open interesting possibilities, although I would tread with caution on versioning voodoo like this :) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog modems behind an Asterisk server?
Hi, -Original Message- Currently, we only transmit at 1200bps, is this rate problematic with Digium cards? Up to what data transmission rate are Digium cards known to work reliable? We do not think we'll ever go beyond 9600bps, can we do this with a let's say TDM400P? On a pure TDM path this should be fine. In fact I think you should not have any real limitation if you set everything correctly. Using VoIP in parts of the link does limit the connection, although we have seen 14k4 connections run stable for a long time. YMMV. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and 2 line MGCP phone
Hi, -Original Message- Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Theoretically you could differentiate by the line: aaln/[EMAIL PROTECTED] aaln/[EMAIL PROTECTED] Are typical indications for this. I've never seen a phone that does this, though.. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] #(transfer) no longer working
Hi Michiel, -Original Message- Anyone who can help me with this ? I tried everything :( exten = s,4,Dial(Local/[EMAIL PROTECTED],5,tTr) exten = s,5,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],10,tTr) Have you tried using the /n parameter for chan_local ? I've noticed it can make surprising difference in some more complex configurations. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP Forwarding
Hi, -Original Message- SIP Phone (xten) - Linksys - Internet - PIX - Asterisk I can get 5060 working with no prob (PIX has a helper built in) but I need to forward RTP 8000 from my linksys to my SIP phone. Is there anyway around the forward? It would be nice to have multiple phones working but I wont be able too since I have to use a foward. Maybe a stun server somewhere on the internet can help you ? You could even run your own on your remote asterisk server. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ENUM NL dead ?
Hi Michiel, -Original Message- Since you already have done something on this, can you tell us what your plan was? Complex :) ENUM was a part of a larger setup concerning roll-out of voip technology over wireless networks. Do you already have some docs about what to do and why, or do we have to setup something like this ? Motivations can be numerous, everyone needs to decide for themselves. A tool with guidelines (very rudimentary) would be usefull. Maybe it's a good idea to talk about this face to face (or in a conference call with some interested ppl) I have web/mail/dns/sip/iax2 services I can make available for this. Think so. I have made contact with dgtp again, hopefully something will come out of that in the next few days. Let's take this discussion off-list. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Softphone for Linux desktops
Hi, -Original Message- We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do For SIP I love kphone. Nice interface, works simple enough, available in many popular linux distro's. For IAX I'd go with iaxComm. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Useragent
Hi, -Original Message- 1- Anybody implement mgcp useragent in *. Nope. Hasn't been done yet. 2- Where can i get that. Not available in your nearest drugstore. 3- if no then anybody can help me to write it down. Digium ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] error message: INIT: Id s0 respawning too fast:disable for 5 minutes
Hi, -Original Message- I have set up [EMAIL PROTECTED] with Digium TDM400P 2FXO/2FXS. I am unable to seize my trunks from either soft or analog phones. Inbound calls result in answer/disconnection. I see the following error code on my asterisk server INIT: Id s0 respawning too fast: disable for 5 minutes Does anyone have any suggestions for me? I'd really appreciate some help on this. I don't think that's asterisk related at all. Check /etc/inittab to see what 's0' is. My guess is it's a serial port that you have connected to something proprietary. If that is the case, just comment out the s0 line and 'telinit q' should stop the messages. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to handle one incoming call on multiple lines?
Hi, -Original Message- Thanks, but it isn't an option because the Telco is actually connected to a PBX which is connected to Asterisk which should act as a intelligent answering device during non-office hours. The PBX isn't capable of doing this. Any other option? Hmm, this is a bit of a hack, but it might suit your needs: - Make sure each of those lines goes into a different extension or context - Add a delay on each line, like this: exten = line1,1,Do stuff exten = line2,1,Wait(2) exten = line2,1,Do stuff exten = line3,1,Wait(4) exten = line3,1,Do stuff exten = line4,1,Wait(6) exten = line4,1,Do stuff Could this help your case ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ENUM NL dead ?
Hi Michiel, -Original Message- I been searching on the wiki and google for ENUM in NL. All I could find were some docs from the Dutch Financial Department about taskforces and plans. But it all links to dead pages and no-longer-connected phone numbers. Is there anyone who knows some more about Dutch ENUM stuff ? If it's all dead, anyone interested in setting it up together with me ? We have been trying to get it alive but have been unable to do so just by ourselves. Some cooperation with the dutch government (dgtp) and domain registry (sidn) will be required. Care to join forces ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 port BRI options ?
Hi Remco, -Original Message- I am thinking of another solution for fax. I have an * box with one PRI card and I'm thinking of adding a quad BRI card in the same box. A separate box running fasx server software will also be equipped with a BRI card for faxing (I cannot use spandsp for various reasons, one of them being that we must log / register all faxes). By keeping the path all digital I'm hoping to avoid trouble with echo (cancellation), interrupted faxes etc. Anyone have an idea on this? It should be possible that way. A few caveats: - Gain settings (also on your PRI/BRI lines) are crucial for faxing - set it too loud and the handshake will fail, set it too soft and the fax will fail after a certain amount of time. Quality of ATA's is also very important with this. - Make sure your box has enough horsepower to cover all the interrupts! Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to setup Dundi in Asterisk?
Hi Ronald, Citeren Ronald Wiplinger [EMAIL PROTECTED]: I subscribed to the dundi mailing list, but so far I have not got a single message from there. Is there a message archive? Its a rather quiet list, yes. I want to setup DUNDI. I have a peering agreemrent, but what is next? I copied from the wiki all parts, but still I am a little bit lost. Has anybody setup DUNDI? We have, ofcourse. I have been out of office these last few days but I will get back to you on your mail, promise :) -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID
Hi, Citeren Anton Krall [EMAIL PROTECTED]: Let mek now what you need Florian and Ill send it offlist. | Seems to me Im been displayed both... How can I control it? | |No way to know that without more in-depth knowledge about your |configuration (i.e.dialplan, what channel have you configured |in asttapi etc.) Show me extensions.conf (and possibly files included there) Also please tell me the following - What channel is making the call - What context is that channel making the call in - What channel is receiving the call - What channel do you have configured in ASTTAPI -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID
Hi, -Original Message- Anton Krall wrote: What re you guys doing for windows callerid from Asterisk besides using yac? Any other working software? With ASTTAPI you can see events for your own phone too. http://sourceforge.net/projects/asttapi/ Take a look at this client: http://www.ivrsoft.com/call-alert.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID
Hi, -Original Message- What do you mean With ASTTAPI you can see events for your own phone too. As opposed to having something message you from the dialplan you can make use of the manager events, that's the point I was trying to make. I already have astapi installed .. Have you tried call alert? Does it work as promised? Yes, it works as promised in my setup. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID
Hi, -Original Message- I just tried call alert but something is wrong.. For each call I get I see 2 or 3 events on the callerid.. The first is the actual number that dialed me, then 1 or 2 entries of my own number. Seems astapi or call alert is recognizing my own number is if I called myself on each call. Is this an astapi issue, misconfiguration on my side or call alert problem? This can be a dialplan issue or a call-alert issue. It is higly dependant on which channel you are monitoring and the way callerid is handled throughout your dialplan. I have seen similar things when using TAPIrex instead of Call-Alert. The TAPI messaging layer allows you to peek into a lot of environment things of the call, including the CALLERID and the CALLEDID (subtle difference ;). Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID
Hi, Citeren Anton Krall [EMAIL PROTECTED]: Seems to me Im been displayed both... How can I control it? No way to know that without more in-depth knowledge about your configuration (i.e.dialplan, what channel have you configured in asttapi etc.) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] outlook express intregation
Hi, -Original Message- All of the stuff I've googled for and read on wiki all relate to Outlook. Has anyone been successful in getting Outlook Express to do click to dial? I don't think Outlook Express has any support for that kind of thing at all. No TAPI hooks in there at least as far as I can tell. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI and PRI together possible?
Hi, -Original Message- I tried it first with the bristuff drivers from Junghanns. The BRI card worked fine alone, but as soon as I load the zaptel driver for the TE110P the BRI card says, that the port is down. Is there any stable way, or as someone experience with this two cards in one box? Yes it can be done (at least with 'real' Junghanns QuadBRI cards, I don't know about the BN card, but I suppose it should work). One thing to note is the way in which modules are loaded. Make sure you don't use modprobe in this scenario, but insmod. This is because with modprobe ztcfg is run after loading the first driver module. This can cause some problems. So insmod all modules, then run ztcfg manually (or from your own script). Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI and PRI together possible?
Hi, -Original Message- Yes it can be done (at least with 'real' Junghanns QuadBRI cards, I don't know about the BN card, but I suppose it should work). It is also possible with the Eicon DIVA Server cards (BRI, 4BRI and PRI). The DIVA Server cards don't use Zaptel, they have to use the CAPI driver, ISDN4Linux or mISDN. I'm not sure if that would work. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (OT) Interesting Product Vocera
Hi Steve, -Original Message- Subject: [Asterisk-Users] (OT) Interesting Product Vocera http://www.vocera.com/products/documentation.shtm Anyone have any experience with this? If these things could speak SIP and were half way decent I could see some real value, even if they are kinda startrekkie Yeah, the concept is interesting. However: - You need the application server to do the actual voice recognition (they supply software/hardware for that) - It was not suited for anything other than US English last time I asked them about it Its not cheap either :) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNMP Monitoring
Hi, -Original Message- I use MRTG to graph Active/Configured SIP channels and Active/Total PRI/ZAP channels, but I don't monitor the up/down status. You could probably Any chance you will share the mrtg setup you used for that ? How did you read out asterisk (via manager interface, tailing logfiles, or ... ?) How busy is your setup ? Thanks, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dutch SIP or IAX numbers
Hi, -Original Message- How knows where I can get a Dutchphone number for asterisk? Pilmo is not delivering one for home use. I think you are physically outside the netherlands, right ? Would you care for an 087 number ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
