Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
No need for the pri debug span, the problem is the duration of the tones when using dtmfmode=rfc2833. It is way to short. A lot of IVR's just don't get enough of the tone to work. The info method still has the correct duration. Simple to test just deal another phone and hit keys, you will see what I mean. Martin Pycko wrote: type on your asterisk CLI pri debug span spanno and send the trace of a broken call regards Martin On Mon, 4 Aug 2003, Stefano Finetti wrote: - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 03, 2003 5:52 PM Subject: Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN Are you experiencing it over PRI ? Can you send the pri debug span spanno trace ?Is your asterisk/libpri code very recent ? I'm experiencing this both over a PRI line (E1), with july CVS, and over a normal ISDN BRI line, with latest CVS sources (taken about a week ago). I'v tried to debug both SIP and using messages (/var/log/asterisk/messages) but i found no useful informations. It's quite important to solve this problem 'cause i'm not able to call some *very* important number used for my job (Telecom HelpDesk, and so on). Thanks, -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
Leave off the softkey xml tags, This should get you working You can telnet to the phone and type debug http and you will get better errors. Maik Schmitt wrote: struggling with localization issues (so the script is not German only) took me a week longer than expected. (Did anybody ever get PHP's gettext extension working??) But finally, I've wrapped something up: Hi, I just tried to use it with our 7960 (sip-version). I've set the services_url in SIPDefault.cnf to http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234; It didn't work with ?user=...pin= cause the phone then tried to get index.php?user=1234\x9fpin=1234?name=SIP... Now I only get: --- CMXML Error XML Parse Error --- When I open the URL with my Browser I see the VoiceMail - INBOX and the source also looks OK: --- CiscoIPPhoneText TitleVoiceMail - INBOX/Title Prompt/Prompt Text^MEs sind keine Nachrichten vorhanden./Text SoftKeyItem NameBeenden/Name URLSoftKey:Exit/URL Position3/Position /SoftKeyItem SoftKeyItem NameOrdner.../Name URLhttp://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234amp;folder=INBOXamp;do=chfolder/URL Position4/Position /SoftKeyItem /CiscoIPPhoneText --- Any Ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fair comparison
Big issues for sip: (Please note I use both Asterisk and Vocal between the two you can have a fairly scalable sip environment with a fair amount of call features.) Pluses for Vocal: For sip switching Vocal is much more scalable, You can have a cluster of UserAgents and Gateways. It never terminates rtp streams so Vocal can not easily be over run with calls. But vocal is mostly just a voip call switch. (Like SER) Negatives for Vocal: Has Zero usable call features, It can route sip calls all day no problem, Don't even try to have it do call features everyone of them has some problem or another. Asterisk pluses: It has call features, Not always implemented the best way but has them in boat loads! Asterisk is an ok switch for sip calls, but you can never have more then one box doing the job. Asterisk Negatives: It crashes. (It is development code) It terminates every sip call that goes through it so can only scale to the point of the boxes ability to excepts the rtp streams. (You can do some clustering of dial plans but this does not help with incoming sip registration and call paths IE your call drops if your box reboots) You may also want to through SER in your list of systems to evaluate. Kim C. Callis wrote: I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? Practical Voice Over IP using VOCAL published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant. Maybe we should try to put together an unbiased comparison (read that as pro/con). I was talking at a meeting about Asterisk, and someone attempted to start flaming Asterisk, and swearing by VOCAL, while another was babbling about the wonders of Bayonne. The only thing that was successful in that meeting about VOIP solutions was tabling that discussion until a future (as in way, way in the future) date. Just a thought! Kim C. Callis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of localecho, questions about call transfers
Dave Alan Caruana wrote: hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? You hit *8# and you will pick up any call you have setup callgroups for. sip.conf: [6004] type=friend username=6004 canreinvite=no callgroup=1 pickupgroup=1 host=dynamic [6003] type=friend username=6003 canreinvite=no callgroup=1 pickupgroup=1 host=dynamic 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. extensions.conf: [local] include = parkedcalls [default] exten = 701,1,ParkedCall(701) exten = 702,1,ParkedCall(702) exten = 703,1,ParkedCall(703) exten = 704,1,ParkedCall(704) exten = 705,1,ParkedCall(705) exten = 706,1,ParkedCall(706) exten = 707,1,ParkedCall(707) exten = 708,1,ParkedCall(708) exten = 709,1,ParkedCall(709) exten = 710,1,ParkedCall(710) parking.conf: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 45 can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
I also had the same problem with sip, I also moved back a couple of weeks in cvs. I also use a AS5300 Cisco in my call chain. I got a bunch of Ignoring this request in debug. I have not had time to trace the call path on this problem yet. Low, Adam wrote: All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP- Asterisk -SIP- AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: 611012210 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI Disconnected from Asterisk server * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Also using the -p (real time option) to start asterisk, may also help. James Taylor wrote: Carlos, You may have something here. Dan - you might try to connect via the virtual network adapter to your host machine's hard drive. (just map a drive) it could be that networking from your VM to the host is faster. James Taylor -- Original Message -- From: Carlos Eduardo Cremon [EMAIL PROTECTED] Dan, What type of virtual harddisk did you create in your linux vmware virtual machine? As far as I know, the default type is a kind of compressed harddisk image, to save disk space in the host machine. Perhaps that is the answer to have bad performance only in local services, opening local files in vmware: the overhead to decompress. -Eduardo Dan escreveu: Nice! If it can work without connecting it to the power supply, then will be better..;-) When we leave the home, I must disconnect from the mains EVERYTHING (the alarm system and the HA PC are the only accepted exceptions)... :-))) Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 1:31 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Hey Dan, Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never know! I can't wait until someone builds one looking like a shoe or a handbag and then I can have them all over the house and the more I have, the happier the other half will be!! W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 11:12 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] Budgetone and Voicemail
Yes I have seen it. I had to change the digit time in the voicemail app and recompile. There is a new voicemail2 app. I have not used it, but maybe it fixes this problem. If you test it out, let me know how it works for you. Brian Borders wrote: I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WILDCARD TDM400P or four Wildcard X100P
Can I use a WILDCARD TDM400P to connect to four Telco circuits aka FXO? Or will I need four Wildcard X100P? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] will this machine handle it
What does -p do? Mark Spencer wrote: You can try using the -p option to Asterisk. Mark On Wed, 2 Apr 2003, Jeff McClure wrote: Good points. This system currently does not use any SIP or IAX channels (or any other form of VoIP) and only deals with 1 call at a time (the single FXO channel is the only link to the outside). At some point, I may add an IAX link to a friend's * box, but I really don't see this setup ever having to deal with more than 2 concurrent calls (one over IAX, possibly with GSM, the other using just the Zaptel channels with no compression at all). Oh...and voicemail uses GSM, of course. The current sound quality is pretty good, but what I do hear are tiny little hiccups during GSM playback in voicemail. Again, I suspect what I'm hearing is the effect of the load on box spiking due to other processes. Does that sound reasonable? Maybe that extra level of detail can help some folks form opinions about required CPU horsepower. --On Wednesday, April 02, 2003 6:43 AM + WipeOut . [EMAIL PROTECTED] wrote: Hi Jeff.. What you are asking is a little bit of a grey area because there are a number of factors that will affect how well you system will perform.. things like the average number of concurrent calls?, are you using VoIP?, what codecs are you using for the SIP of IAX channels? and no doubt a few others.. But here is my experience.. I am using a PII-400 and with 2 concurrent VoIP calls using G.711 codec the processor barely registers anything.. So I should thing that this system should handle 10-15 concurrent calls... If I used the GSM codec for example I an sure this number would drop significantly.. Hope that helps.. Hi folks, Right now I'm running * along with a lot of other apps on my firewall box, which is a P-II 400 with 192MB of RAM. I have a single T100P card connected to a channel bank that's using one FXO and two FXS ports. I want to move * off to another computer (mostly because I think the other apps on the current box are causing enough of a load to affect the sound quality a bit). I'm looking for a computer to put it on, and I've found someone with a P-II 350 with 64MB of RAM (I could steal another 64MB from the firewall if I have to). So, I need an opinion from some more experienced users. Given the same number of ports and assuming I don't run any other apps on the box, is that P-II 350 beefy enough to handle my * setup comfortably? Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Jeff McClure [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect and incoming DTMF
The pending route patch may fix you (I'll be testing it once it is merged) but if not, try using a b2bua on out going calls this should hide Asterisks ugliness from Iconnect. A mostly working b2bua can be found here: http://www.vovida.org/downloads/vocal/1.5.0/rh73/b2bua-1.5.0-20.i386.rpm I don't connect to Iconnect, but A b2bua does allow me to interface Asterisk with other SIP environments that it would not other wise be able to do. DA wrote: No response on this post from a couple of weeks ago, so I thought I'd send it out again. Any thoughts on DTMF on inbound Iconnecthere calls? Does anyone have this working? Thanks again. --- DA [EMAIL PROTECTED] wrote: It would be awesome to have incoming DTMF work with Iconnecthere. Has this functionality been added to Asterisk? If yes, does anyone have a working config to share? Thanks a bunch, DA --- Matthew Farley [EMAIL PROTECTED] wrote: I now have (most) of the bugs worked out of my SIP-only asterisk installation, but one fairly serious issue remains. I am using the dtmfmode=inband in both my general area as well as the extension-specific area (for iconnect) in sip.conf, but only DTMF only works on calls placed out through iconnect to the PSTN. DTMF tones coming from the PSTN into asterisk through iconnect are not recognized. Does anyone have a working setup where they dial into asterisk via iconnect (from a PSTN phone) with DTMF working? Any suggestions as to how I can get this to work would be greatly appreciated. I would really like folks to be able to dial into this system from outside to check voice mail and such, but without DTMF recognition on those calls, I see no way to accomplish this. Thanks! -- Matthew Farley [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Platinum - Watch CBS' NCAA March Madness, live on your desktop! http://platinum.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Tax Center - File online, calculators, forms, and more http://tax.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -p
Thanks I tried --help, -h is a little non standard for GPL software. -h normally means human readable. See ls, df...ect But what ever works! This does fix a wold of problems. It really should be on by default, with an option to turn it off, not on. smile Happily added to my init scripts for Asterisk. Just my opinions of course YMMV. Andre Bierwirth wrote: Asterisk CVS-03/31/03-00:53:00, Copyright (C) 2000-2002, Digium. Usage: asterisk [OPTIONS] Valid Options: -h This help screen -r Connect to Asterisk on this machine -f Do not fork -n Disable console colorization -p Run as pseudo-realtime thread -v Increase verbosity (multiple v's = more verbose) -q Quiet mode (supress output) -g Dump core in case of a crash -x cmd Execute command cmd (only valid with -r) -i Initializie crypto keys at startup -c Provide console CLI -d Enable extra debugging -p = Realtime Priority Andre - Original Message - From: James Sizemore [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 05, 2003 12:26 PM Subject: Re: [Asterisk-Users] will this machine handle it What does -p do? Mark Spencer wrote: You can try using the -p option to Asterisk. Mark On Wed, 2 Apr 2003, Jeff McClure wrote: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and rfc2833 dtmf + # transfer
Yes. Lele Forzani wrote: Has anybody noticed that # transfers aren't working anymore when SIP is used with rfc2833 dtmfmode? They work as espected with inband dtmf. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Gui for Asterisk (AGI questions)
I agree, whole heartily, No XML please! I suggest the requester, take a look at Vocal if he thinks XML is a good ideal for any-e-thing at all. I am glad most Unix configuration files have avoided XML hell. Problem will all XML configs: 1. They are nearly imposable for a human to read, for any non trivial config. 2. Thus requiring a XML app to edit the config. Which in every case I have ever seen is always out of date with the options that are needed. Requiring some poor sap to read though thirty pages of crap to edit some XML tag to turn on some option, that the edit app does not yet support. 3. I personal find it easer to parse a human readable config file, then deal with any XML library that I have ever seen. 4. Seeing as you need to agree what tags you read in from application to application, what the hell is the difference from creating your own config format then your own XML format? Take XML bookmark files. Every web browser has one. And no other web browser will read in any others bookmark file. Galeon and Konqueror will read each others XML bookmark files but they collaborated on the format! And could just as well have collaborated on a flat text file that was easy for a human to read! 5. So what was the point of XML again? They is none! Mark Spencer wrote: Would someone like to propose what an XML extensions.conf would look like? How about an XML zapata.conf? I know XML is a fun buzzword and as a syntactic hammer seems instantly appropriate for every configuration nail, but I think in practicality, XML does not lend itself to describing things like zap interfaces as easily and certainly not as compactly as the existing syntax does. Sure, it takes about 5-10 minutes to understand the nature of Asterisk's config files, but the time is well worth it, and you'll understand why it is done the way it is. Why incur the overhead of trying to parse XML? Mark On Mon, 17 Mar 2003, Chris Albertson wrote: This topic is of interrest to me because I have to re-write the conf. file system on some software I'm working on. It's currently horible. (Just keyword=value pairs minus the keyword= part) SOAP looks to me like a message passing protocol. Configuration needs to be placed in a persistent storage like a file. Sometimes db tables, LDAP, or a DBMS is used. Either way it's storage SOAP looks like a way to send messages, not a way to store data. But SOAP is XML, So I'm glad you agree about the part. --- Jeremy McNamara [EMAIL PROTECTED] wrote: SOAP My 2 cents, Jeremy Chris Albertson wrote: I think the way to go with conf. file for Asterisk is XML. When I first saw the Asterisk conf files I wondered if Eric Allman had found a new job working on Asterisk. (That's a joke for those of you who have had to maintain a sendmail installation. sendmail.cf is the definition of cryptic) Some advantages of XML: 1) Parsers and file editors already exist for XML. Users could edit files with ready made GUI tools, programmers can use XML with XML libraries. There are even web-based tools for maintaining XML data. 2) Parsers and file editors can perform file validation. Making it not-possible to save an invalid file. 3) (some) Database systems can gobble up XML and spit it back out. Yes, I think the DBMS idea was resonable for a large installation. Overkill if less then say a few hundred extensions. Large sites like to manage phone extension and, extension to physical location maping and other stuff in a DBMS. 4) XML (with addition of a style sheet) can be directly displayed in a web browser 5) Without a GUI and/or wrb front end the system will remain only geek usable. (Your average phone guy doesn't know how to use vi.) 6) XML readers can ignor parts of the XML file they don't understand. This allows one file to carry information for multiple readers ad for new additions too the file not to break older readers. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2003-03-17 at 11:36, Stefano Finetti wrote: I was wondering about a little php-based GUI to manage Asterisk Extensions. Many way to obtain this, but i think that implementing in a php script the AGI Commands should obtain the best results (more, the best result would come with AGI+Mysql instead of a text file like extensions.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Gui for Asterisk (AGI questions)
Chris you seem to like the things I hate the most! LOL About the only thing I hate worse then XML for config files is using M4 for any-e-thing at all!!! grin Your a sick man, you just seem to love needless steps in editing a config file! I'll stick to Macros, myself. Chris Albertson wrote: Parcers running even on older PC hardware can parce XML faster than the characters can pass under the read/write head of the disk. The speed difference between XML and non-XML is due only t the extra character count. Modern parcers are very good. CPU cycles are not a good argument. Where XML is usfull is where you have several programs that need to read the same data files and do different things with the same data. And the data is non-triveal complexity And the programs that read the data are maintained by different organizations. XML has pretty much taken over B2B e-commerce. XML formatted invoices and so on... One program might apply a style to the invoice so it can be printed an other might log it to a database Ok, I admit .conf files that descibe telephony hardware connected to a PC don't need to be XML. What else but Asterisk would read these files? The dail plan is different. IMO data about users, their phone extension(s) physical location, name, authenication info (passwords) and other personal data needs to go in an on-line data storage system. Call it a RDBMS, LDAP server or even an NIS map. Sepporate from the above is the flow of control that a call takes. (ring phone(s), answer or go to voice mail or secritary) You should only have to define how a type-D' phone flows once then in the above on-line database simple note that the extension is of type-D extensions.conf combines the above two types of information. A typical medium sized company would have many repeted blocks difering only by extension number in extensions.conf The best thing might be to seporate the two types of data. The simple thing to do is use a preprocessor like M4. Defin a macro for a type-D phone and then have lines like extn-type_D(6578) extn-type_D(6579) Then to change the flw of control for all 100 extension you only have to change the macro definition. Still you are keeping two diferent kinds of data in the same file but at least you've factored out the redundent information --- Peter Brown [EMAIL PROTECTED] wrote: Hi, XML may be the latest but it also adds latency to the whole process - for what benefit? It looks better, we are using the latest technology? If a wheel barrow will do the job why get a D9 Tractor? No flame wars pls, just my 2cents worth. Peter At 19:00 17/03/2003 -0500, you wrote: I hate to say do it Microsoft's way; but they FINALLY came around with Win2003 to storing the web server config in XML; and after revisions of registry storage (basically param=value format), then metabase with inheritance issues (custom format, no tools to edit) and now they went XML. I've always liked the apache layout (although I make a living on IIS) - This new XML one, although I haven't played with it much yet, looks like the way *ALL* configs should be. Not that IIS config is the way - but XML. As was said, other editors can do it, there's components (windows and *nix based) to parse xml readily available, etc. I've said for a long time xml is NOT the be all and end all like people profess, and it's ended up doing things that there's no reason to do - however for config files it looks like a great answer. Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers BitShop, Inc. - http://www.bitshop.com - $149/month colo special -Original Message- From: Chris Albertson [mailto:[EMAIL PROTECTED] Sent: Monday, March 17, 2003 5:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PHP Gui for Asterisk (AGI questions) I think the way to go with conf. file for Asterisk is XML. When I first saw the Asterisk conf files I wondered if Eric Allman had found a new job working on Asterisk. (That's a joke for those of you who have had to maintain a sendmail installation. sendmail.cf is the definition of cryptic) Some advantages of XML: 1) Parsers and file editors already exist for XML. Users could edit files with ready made GUI tools, programmers can use XML with XML libraries. There are even web-based tools for maintaining XML data. 2) Parsers and file editors can perform file validation. Making it not-possible to save an invalid file. 3) (some) Database systems can gobble up XML and spit it back out. Yes, I think the DBMS idea was resonable for a large installation. Overkill if less then say a few hundred extensions. Large sites like to manage phone extension and, extension to physical location maping and other stuff in a DBMS. 4) XML (with addition of a style sheet) can be directly displayed in a web browser 5) Without a GUI and/or wrb front end the system will remain
Re: [Asterisk-Users] ParkedCall and SIP.
I got some time this week end to play with this. By add the pickup lines in extensions.conf: ; ;Parked calls ; extern = 701,1,ParkedCall(701) extern = 702,1,ParkedCall(702) extern = 703,1,ParkedCall(703) extern = 704,1,ParkedCall(704) extern = 705,1,ParkedCall(705) extern = 706,1,ParkedCall(706) extern = 707,1,ParkedCall(707) extern = 708,1,ParkedCall(708) extern = 709,1,ParkedCall(709) extern = 710,1,ParkedCall(710) I got parked calls to work with SIP. The instruction seem to indicate I would not need the last step. shrug James O. Sizemore III wrote: I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext = 700 parkpos = 701-710 context = parkedcalls ;parkingtime = 45 /etc/asterisk/extensions.conf include = parkedcalls include = default [default] extern = 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and SIP. To use it I should be able to hit # then get a prompt? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ParkedCall and SIP.
Thanks, P.S. What is the URL to your wish list? smile Mark Spencer wrote: SIP does not yet support parking unless you do #transfer support. The reason is that once you have done a transfer in SIP, the original call is gone, so there is no way to announce where the call has been parked. Mark On Fri, 7 Mar 2003, James O. Sizemore III wrote: I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext = 8540 parkpos = 8541-8555 context = parkedcalls parkingtime = 45 /etc/asterisk/extensions.conf include = parkedcalls include = default [default] exten = 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and SIP. To use it I should be able to hit # then get a prompt? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
I send example XML files a week or so ago, as well as an impotent line for you dhcp server. Mike Reiling wrote: Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and MWI 7960
I don't think you understood my reply, I also don't use Asterisk as a UA no calls are registered to Asterisk. Asterisk just needs to be able reach the phone. For this you need a peer statement for each phone. That never takes calls. And a way for Asterisk to reach the phone when it has a MWI message (host=phone_ip). You can supply this ip with DDNS records. Asterisk needs the ip supplied to it some-way, if the phone registers with asterisk it has the ip to send the MWI message. If the phones does not register, you need some way to supply the ip. With out DDNS you would not be able to move the phones from LAN to LAN and still get MWI. Again you will need a peer like this: [2114] type=peer username=2114 insecure=yes canreinvite=no context=default mailbox=2114 host=SIP003094C274B3.bna01.isdn.net For each phone. billp wrote: I still do not think you understand my question... I am using 'ser' for my SIP gatekeeper. I am using asterisk for VOICEMAIL ONLY. The only time incoming calls touch asterisk is when someone does not answer their phone, and SER (the SIP gatekeeper) redirects the caller to asterisk/port 5110 to take a message. So- the question is- if asterisk is not the main SIP gatekeeper, can it still signal the 7960 phone's MWI? Either way-- I am looking for a successful unsolicited NOTIFY text that actually turns on the MWI on a SIP 7960. From that, we can write our own MWI on/off routines if necessary. thanks bill On Thu, Mar 13, 2003 at 06:47:44AM -0600, James Sizemore wrote: Message waiting indication work fine you just need to set up DDNS for phones Asterisks Needs to know how to reach the phones! [2114] type=peer username=2114 insecure=yes canreinvite=no context=default mailbox=2114 host=SIP003094C274B3.bna01.isdn.net You will need to have no voice mail be for you start to test this. Also your phones control port needs to be 5060, Asterisk will not use another port. billp wrote: Two issues-- is anyone using Asterisk as a gatekeeper with cisco 7960 phones and cisco gateways? Experiences, thoughts, etc appreciated. If anyone has moved from/to ser to/from Asterisk, I would be interested in hearing experiences... We have been trying to get message waiting indication working on our 7960's without luck. Is anyone using MWI on 7960's when Asterisk is ONLY being used for voicemail, and not for a gatekeeper? If anyone has MWI working successfully with 7960's, would it be possible to get a dump of a successful NOTIFY message that turns a light on/off? thanks bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] # Ouch ... error while writing audio data: : Broken pipe
Just check-out asterisk from cvs, It compile but crashes right off with? # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Any ideal how far back I need to go to get a working build? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Ouch ... error while writing audio data: :Broken pipe
Hell rm -rf asterisk cvs checkout asterisk make samples Same thing! Mark Spencer wrote: make clean ; make install? Mark On Sun, 9 Mar 2003, James Sizemore wrote: Just check-out asterisk from cvs, It compile but crashes right off with? # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Any ideal how far back I need to go to get a working build? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
The only problem I can think that you would have with the ztdummy would be that to used a kernel source other then the one your running when you build it... So what errors did you get when you build ztdummy? Rattana BIV wrote: No I use chan_capi and H323 but not zaptel device. So can I use it ? When I lauch ztdummy I have some errors. Regards Rattana - Message d'origine - De : Brancaleoni Matteo [EMAIL PROTECTED] À : [EMAIL PROTECTED] Envoyé : mardi 4 mars 2003 18:32 Objet : Re: [Asterisk-Users] a problem with MeetMe have you any zaptel device in your box? a zaptel device is required for timing source for the conference (so meetme) matteo Il mar, 2003-03-04 alle 16:49, Rattana BIV ha scritto: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
Uncomment ztdummy from zaptel/Makefile make clean ; make install modprobe ztdummy. Restart asterisk, all fixed. Rattana BIV wrote: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice-mail App
A few note about each file. OS79XX.TXT: Should always have this old version of the code for the phone. Ringlist.xml: Lets you have custom ring tones (not a good to have Bart saying eat my shorts as your ringer) SIP-MAC-ADDRESS.cnf : Phone setup goes here. Set You telnet password if you want to log into the phone and do sip debugging. SIPDefault.cnf: Default values, note this is were you tell your phone what version of code to download. dialplan.xml: The dialplan for the phonemostly to set timeouts before dialing. John Todd wrote: I'm interested in the example xml configs, if you have some to send. I'm getting some 7960's shortly, and would appreciate the hints. JT In case you have never setup a 7960 before. The easies way is to setup dhcp and have the code on a tftp server. option tftp-server-name your.tftp.edu; If you need a copy of the xml configs or dial plans for the phone let me know, and I send some your way. There is a nasty bug with REFER in the SIP code for that phone before version P0S3-04-3-00.bin Mark Spencer wrote: I have played with the timeouts :int timeout, int ftimeout for ast_readstring(chan, password, sizeof(password) - 1, 2000, 1, #) To no effect, Could you give me a pointer to where I can start looking next to track down this strangeness. Maybe something in the SIP driver? Getting in and out of band DTMF? etc... Best place to look is probably the RTP code, where the digits are generated. Specifically look at the rfc2833 routines, assuming that's how they're being sent. I just got a 7960 on loan, so I'm going to set it up so that I can try to duplicate any problems you're having. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users xml-7960.tar.gz Description: GNU Zip compressed data
