Re: [Asterisk-Users] incoming calls in h323 do not come to right dialplan
[EMAIL PROTECTED] wrote: Good to hear I am not alone. Actually, I am using the Nufone's h323 module. Still this creates the problem. I had a braod look at the code and it seems that it is not possible that incoming calls go to other places than general context (I am not sure I understood it all, but almost). I had a similar problem, my Siemens OptiPoint 300 A refused to send calls to the corresponding context, so after a short review of the code I found the parameter userbyalias and included userbyalias = no in h323.conf and now chan_h323 use the IP address to match, not the H323_ID. /Soren ___ 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 in HEAD broken?
Bill Seddon wrote: Are there new instructions for compiling the zaptel driver in HEAD? I compiled the zaptel driver from HEAD successfully last weekend but trying to compile the current driver for another machine results in the error: zaptel.c:45:31: linux/moduleparam.h: No such file or directory If I go to compile zaptel on the machine that compiled successfully last weekend, the same error occurs. So far as I can tell, I don't have a file called moduleparam.h anywhere on either machine. Yeah, the culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include linux/moduleparam.h to: #ifdef LINUX26 #include linux/moduleparam.h #endif and now it compiles on 2.4... /Soren ___ 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
[Asterisk-Users] FYI: NIST issues recommendations for secure VOIP
Following is sharelessly copied from one of the newsgroups I read on grc.com.. /Soren NIST issues recommendations for secure VOIP http://www.gcn.com/vol1_no1/daily-updates/34747-1.html http://csrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf *** Quote *** The National Institute of Standards and Technology has offered some cautionary advice for offices considering moving their telephone systems to voice over IP. VOIP introduces both security risks and opportunities, NIST said in a recently released report. Lower cost and greater flexibility are among the promises of VOIP for the enterprise, but VOIP should not be installed without careful consideration of the security problems introduced. The report, Security Considerations for Voice over IP Systems [1], offers recommendations for using VOIP. According to NIST, the need to logically separate voice and data traffic, provide backup power and ensure business continuity could seriously compromise the promise of a single, inexpensive voice and data network. NIST Special Publication 800-58 is the final version of a draft report published in April by the agency's Computer Security Division. [...] *** Unquote *** [1]: http://csrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf ___ 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: 'I'nvalid extension handling problems, even with workaround
;;;extensions.conf
[internal] ;;; context used by our internal SIP-phon
include = voiptalk.org ;include context below
exten = 11,1,Dial(SIP/gsbt100,20,tr);calling : dial our office phone
include = invalid_calls;all ext numbers not handled above are
invalid
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
[invalid_calls] ;;; default context for invalid calls
exten = i,1,Wait(1)
exten = i,2,Answer
exten = i,3,Playback(invalid)
exten = i,4,Hangup
;;;end of extensions.conf
More here: http://www.voip-info.org/wiki-Asterisk+i+extension
/Soren
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Re: [Asterisk-Users] Aterisk@Home
Walid Azab wrote: Hi All, We have just installed [EMAIL PROTECTED] It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated. http://your_server/maint /Soren ___ 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] h.323 Type=User
Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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] TDM400P FXS polarity reversal?
Andrew Kohlsmith wrote: On December 13, 2004 03:10 am, Soren Rathje wrote: wait_just_a_bit(HZ/10); I didn't want to wait inside the driver, likely a place where interrupts are disabled... Well, nobody claimed it was ready for production.. :-) I'm usually OK for POC code, but don't ask me to do production code, I haven't done any serious coding the last ~25 years. I usually tell programmers what I want and how I want it and where to find the bugs... /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] CVS zaptel missing files
Greg - Cirelle Enterprises wrote: it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c It was renamed to wctdm.c around Nov. 6. 2004 /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TDM400P FXS polarity reversal?
Something like this.. :-)
reg72 = wctdm_getreg(wc, card, 72);
/* Negative Voltage */
if (reg72 6) {
wctdm_setreg(wc, card, 72 , reg72 ^ 0x40);
wait_just_a_bit(HZ/10);
wctdm_setreg(wc, card, 72 , reg72 0x3F);
}
/* Positive Voltage */
else {
wctdm_setreg(wc, card, 72 , reg72 0x3F);
wait_just_a_bit(HZ/10);
wctdm_setreg(wc, card, 72 , reg72 ^ 0x40);
}
I just haven't found the place to put it..
/Soren
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Re: [Asterisk-Users] How to Playback Mailbox Owners Name?
Thorben G. Jensen wrote:
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but + Mailbox Owner Name +
has gone to lunch
Extension 999 in voicemail context internal
exten = 999,1,SetVar(VM_CONTEXT=internal)
exten = 999,2,Playback(im-sorry)
exten =
999,3,Playback(/var/spool/asterisk/voicemail/${VM_CONTEXT}/${EXTEN}/greet)
exten = 999,4,Playback(flagged-for-lea)
exten = 999,5,Hangup()
That should get you going.. :-)
/Sren
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?
Soren Rathje wrote: Specs for Si3210 (TDM400P FXS Module) says on page 93: --- Register 72. On-Hook Line Voltage Bit 6 VSGN On-Hook Line Voltage. The value written to this bit sets the on-hook line voltage polarity (VTIPVRING). 0 = VTIPVRING is positive 1 = VTIPVRING is negative --- The (missing) link.. https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/ProSLIC/en/si3210.pdf Features ... ... * Software programmable signal generation and audio processing: - DTMF generation and decoding - 12 kHz/16 kHz pulse metering generation - Phase-continuous FSK (caller ID) generation - Dual audio tone generators - Smooth and abrupt polarity reversal == NB! - -Law/A-Law and 16-bit linear PCM audio ... ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TDM400P FXS polarity reversal?
Rich Adamson wrote: Is it possible to do a talk battery polarity reversal on a TDM400P FXS interface? Everything I can find seems to be referring to the procedure for detecting a battery reversal on a telephone company POTS line using the FXO interface, but not for actually generating one back to a station upon answer supervision. I would assume that VoicePulse and VoipJet provide a way of signaling far-end supervision back to the originating Asterisk PBX... Basically, my two questions are: (1) Is the hardware capable of even performing a reversal? (2) If the above is true, how would you make it happen in Asterisk? The Silicon Labs spec sheet does not specifically indicate generating a reversal is possible. Therefore, best guess is the integrated circuits on the card does not support it, therefore asterisk has no means of doing it. Specs for Si3210 (TDM400P FXS Module) says on page 93: --- Register 72. On-Hook Line Voltage Bit 6 VSGN On-Hook Line Voltage. The value written to this bit sets the on-hook line voltage polarity (VTIPVRING). 0 = VTIPVRING is positive 1 = VTIPVRING is negative --- I wonder if this can control Pol-Rev's.. ? /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] long list of prefixes
Randy Bush wrote: if a phone number starts with one of 50+ prefixes, i want to send the sip call to gateway X. if it is in any other prefix, i want to send it to gate Y. Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc I run a home server so I have never had the need to do stuff like that but it looks like the thing you want. I'm sure there are other alternatives out there that will do the same only differently... /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X100P does not detect ringing
Michael Vogel wrote: Hi! The X100P is working - partly. I can make outgoing calls. But the card has got a problem detecting incoming calls. Even in verbose mode I don't see any hint that the card detects a call. At the moment I'm using the card behind my old ISDN telephone system. Could it be a problem with the line voltage? Can I adjust anything when loading the kernel module? I presume it is a clone card you have, it behaves differently when having opermode=1 set. The original X100P uses linear (DC) termination and the clone, when set to CTR21, uses complex (AC+DC) termination, this means the signal gain is not comparable. The current wcfxo driver is NOT using any of the registers to determine RING/OFFHOOK/etc., these indications are found by listening to the line so the rxgain/txgain settings in zapata.conf does matter. By using ztmonitor (described here http://www.voip-info.org/wiki-Asterisk+x100p+echotraining) you can train the ears of Asterisk. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Which modem is known to work with asterisk?
Michael Vogel wrote: Thanks for the both eBay-links. At first I try a cheap oem x100p cause the dealer assured me it will run in germany and with asterisk. remember to add the option opermode=1 to your wcfxo driver to switch the card to CTR21 termination mode, otherwise it defaults to FCC termination mode (DC 600 ohms) and you have gained nothing. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Which modem is known to work with asterisk?
Diego Aguirre wrote: Hi, Intel Modems based on chipset Ambient MD3200 works fine! WinModem's labelled AMI-IE92 using the Ambient MD3200 global chipset (Si3021, Si3014) work fine and are CE/RTTE approved -- WITHOUT SUPPORT FROM DIGIUM. Note: The Wildcard X100P/X101P only have FCC approval. Alternatively the TDM400P is using a global chipset -- WITH SUPPORT FROM DIGIUM. Follow your conscience :-) /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] I Am Missing Something Somewhere Somehow!
Adnan Ahmed wrote: hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. Please post console log with errormessage.. My guess is the host=192.168.10.195 definition and the use of context=sip not matching the dialplan. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Which modem is known to work with asterisk?
Michael Vogel wrote: Soren Rathje schrieb: Note: The Wildcard X100P/X101P only have FCC approval. What does that mean for me? Is it illegal to use it in germany or do they don't work in germany? They will *probably* work in Germany.. Check with T if they allow non-CE approved equipment, not that I think you will go to jail but in case of problems they will throw the blame at you regardless.. Alternatively the TDM400P is using a global chipset -- WITH SUPPORT FROM DIGIUM. Hmm, the TDM400P is a little bit more expensive than the X100P, isn't it? Well, if you buy new brakes for your car on eBay.de they will work but you get no guarantee, if you buy them in the autoshop you will get 2 years guarantee but they will cost you more... Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Which modem is known to work with asterisk?
Michael Vogel wrote: [snip] The question for me is, if I buy this card for a little money or if I buy an external modem with voice chipset. There _should_ exist some. But I don't know which ones. If the pictures show the true card Check first!!! I've seen this so many times before - old picture, new card, different chipset. http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemcategory=8057item=6724669731rd=1 http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemcategory=8057item=6723748329rd=1 /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Siemens optiPoint 300
Ed Greenberg wrote: Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to them in the archives of this list, and the Wiki seems to be down. I have a chance to pick up a bunch of these, cheap. Questions: * Asterisk support? * What sort of power supplies will they need? The bunch I am looking at are surplus and have no supplies. I have a few OptiPoint 300 Advanced that I've played with but they are not a priority at the moment. chan_h323/CVS-HEAD works without callerid. Only the basics... chan_oh323 - not heavily tested... When I get the time I'll try with a Gatekeeper to see if that changes anything.. Powersupply Type: FW 6798 (Made in Germany) Input: 230V~/50Hz/73mA/16,8VA Output: 19V~/600mA/11,4VA Plug (phone): RJ12 pin 1+6 /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 process
David Boyd wrote: [snip] Greg, you need chill; take a deep breath; now say to yourself, let it g!! Does hypertensio arteriale and myocardial infarction ring a bell..? Critch, has the right to respond, anyway he desires. People need to be responsible for themselves and their actions, and in particular they need to defend themselves if they feel attacked or insulted. I have not seen a response from the individual who posted the original question (Hong) reply at all to the thread; if he isn't concerned then why are you? He probably resigned from the maillinglist screaming... Why do you think the list as a whole reflects something about you, only your posts say anything about you. I don't wonder at all about Linux catching on, it is, one informed user at a time! Can we conclude that the following chinese proverb still is valid ?? He who asks may be a fool for five minutes. But he who does not ask remains a fool forever. On the other hand... This may be more appropriate... Accept that some days you're the pigeon, and some days you're the statue. Scott Adams. :-) /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FXO setup
Lex Lethol wrote: So, if zaptel will not read codes from my indications conf file, what would be a suitable solution to feed it my country tones? Is there a list of loadzone/defaultzone country codes included in zonedata.c? Can this list be updated to include my country? (I have the tones) Take a look at zonedata.c, it should be obvious to you as you already worked on the indications.conf file. Once you recompile Zaptel and everything works you can submit the changes to bugs.digium.com to make the internationalisation more complete. :-) /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 11:46:15AM +0100, Soren Rathje wrote: Can you post your actual configuration ? /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) /etc/asterisk/zapata.conf [trunkgroups] [channels] context=default Remove the next two lines, switchtype is related to ISDN circuits and the signalling you specify later in the file. switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes ; Type of caller ID signalling in use ; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands ; cidsignalling=bell ; ; What signals the start of caller ID ; ring = a ring signals the start, polarity = polarity reversal signals the start ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=internal callerid=House Phone 1234 # Change to match your settings.. signalling=fxo_ls You may want to change Loop Start to Kewl Start (fxo_ks) channel=1 context=incoming callerid=asreceived signalling=fxs_ls You may want to change Loop Start to Kewl Start (fxs_ks) channel=2 /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: Hi Cirelle, On Sun, Nov 14, 2004 at 07:28:56AM -0500, Cirelle Enterprises wrote: you might have to power the box down - no power for the modules to load (appears to be common for this card) if that is the case, do a search on tdm in the email archive as there is a fix for the reboot problem I'm afraid I don't understand this sentence. I did power the machine down to make sure that the power was indeed plugged in. Are you saying there's a startup problem with these cards. module 1 (closest to the top of the bracket (furthest from the pci connector) is for the phone line module 2 is for the handset I don't agree, but perhaps I'm wrong. The green module (fxs) is number 1 and the red module (fxo) is number 2. As I understand it you plug the handset into the green one (fxs). Not so ? Green is phone Red is line if we're using modem terminology.. :-) /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 02:22:18PM +0100, Soren Rathje wrote: /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) Thanks Soren. I made all the changes you suggested, but do I have to change the above to ... fxoks=1 fxsks=2 ... if I changed to kewel-start in zapata.conf ? My fault, you should change it in both files. I assumed so, and went ahead and did so. Still no dial-tone though. Hmm.. Does Asterisk load chan_zap ? /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 03:01:28PM +0100, Soren Rathje wrote: Hmm.. Does Asterisk load chan_zap ? I believe so: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXO Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote: Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. It's the middle pair. I assume that's 2+3 on an RJ connector ? Correct.. Just for verification, do you have any green led's lit on the back of your card ?? /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: I think I know what the problem is. I think that asterisk cannot generate dialtone because it had a problem with the soundcard. [chan_oss.so] = (OSS Console Channel Driver) Nov 14 16:35:49 WARNING[2078]: chan_oss.c:994 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Nov 14 16:35:49 WARNING[2091]: chan_oss.c:240 sound_thread: Read error on sound device: Resource temporarily unavailable I had to put this in modules.conf to get rid of the error: noload = chan_oss.so So I assume now that it's not capable of making dialtone ? I have noload'ed both chan_oss and chan_alsa and I still get a dialtone. OK, Excercise 1; Stop Asterisk, Stop Zaptel. (I'm using FC1 so do the equivalent for Debian) modprobe zaptel debug=1 insmod wctdm debug=1 /sbin/ztcfg Now you can tail -f /var/log/messages and see hookstate. Already at this point I get a dialtone on my FXS port. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: kernel: NO BATTERY on 1/2! I don't like the look of that NO BATTERY message. What do you think Soren ? NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) BTW. Does the hookstate change change is you lift the handset ?? /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote: NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) ok. I connected it to the PABX and I got this so I assume that ports ok Nov 14 17:49:30 zorg kernel: 63761 Polarity reversed (0 - -1) Nov 14 17:49:30 zorg kernel: BATTERY on 1/2 (-)! BTW. Does the hookstate change change is you lift the handset ?? In /var/log/messages ? nothing happens as far as I can see. May I suggest you call the nearest medicine man and have him drive out the gremlin... Or, look for contact problems in the sockets/connectors, you may have a faulty FXS module since the FXO module and the base card seems to function as expected. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] (newbie) no dialtone on a TDM400P card
Thomas Andrews wrote: I've never set up asterisk before. I hear no dialtone on the telephone plugged into the TDM400 card. This is what ztcfg -vv gives me: Zaptel Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXS Loopstart (Default) (Slaves: 02) 2 channels configured. I've used this link to set up the ports: http://www.digium.com/index.php?menu=faq#Configuration_0 Can you post your actual configuration ? /etc/zaptel.conf /etc/asterisk/zapata.conf /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 - UK Impedance problem with X100P?
Rich Adamson wrote: [snip] The x100p (and presumably the clones) have an integrated circuit on the board that was manufactured for use in the US with 600 ohm pstn lines. The chip cannot be changed to any other impedance. However, there can be many different sources for the echo and impedance matching is only one of them. Others include: - incorrect * zapata.conf parameters - poorly engineered motherboards (eg, poor PCI bus, interrupt latency) [snip] There are two types of alternative cards (not starting a discussion on who made it first!), one called AMI-IA92 (Si3021/Si3012/MD3200) with a FCC approval and one called AMI-IE92 (Si3021/Si3014/MD3200) with a RTTE approval. The AMI-IE92 have additional components on board for setting AC real/complex termination as well as DC termination (Japan Mode/FCC Mode/CTR21 Mode). Both card types are compatible but the wcfxo driver does not utilise the additional settings on the AMI-IE92 card. Source: Chipsets (login required) https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/Silicon_DAA/en/si3034.pdf https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/Silicon_DAA/en/si3035.pdf Amigo Communications product sheet (www.amigocom.com) http://61.31.72.100/amigocom/products/ami_ia92_ie92.htm /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 - UK Impedance problem with X100P?
Rich Adamson wrote: [snip] For those that would like to play around with the above, might take a look at zaptel/fxstest.c (and the associated Makefile complile options) as it can be used to modify/view the tdm04b chip parameters. I'm not a programmer, but doubt whether it would take much to modify it to exercise the cards noted above. [snip] UK is a CTR21 country and after having a closer look at the wcfxo.c code it is supported *if* the card have the global chipset (Clone only, I believe). To enable CTR21 you have to modprobe/loadmod/whatever the wcfxo driver with the parameter 'opermode=1'. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AST doesn't start after update from 0.5 to 1.0
Joerg Beck wrote: [CUT] Warning: phx.c:2304 ast_register'_application: Already hvae an application 'Voicemail2' Warning: phx.c:2304 ast_register'_application: Already hvae an application 'VoicemailMain2' Warning: loader.c:334 ast_load_resource: app_voicemail.so: load_module failed, returning -1 Warning: loader.c:429 load_modules: loadin module app_voicemail.so failed! My guess is that you did not clear out /usr/lib/asterisk/modules and that you have some old modules in there playing tricks with you. Check the date on the xxx.so files. Optionally you can noload = xxx the old modules in /etc/asterisk/modules.conf if you do not want to delete them. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] first tries !
[EMAIL PROTECTED] wrote: hmm... now tring.. somone to know how can I redirect the output of the sip debug into file 'cause it is really hard to grasp (several pages is just one call) Try this.. http://www.voip-info.org/wiki-Asterisk+debugging QUOTE If you are having problems catching intermittent problems on your system, consider adding more information to the Asterisk message log. If you look in logger.conf you will see something like: messages = notice,warning,error consider changing this to: messages = notice,warning,error,debug,verbose (for short periods of time anyway it can really eat disk space). Note also (AFAIK) you will need to restart Asterisk to get this change to take effect (reload doesn't do it) If you really want to see what's going on under the hood you can: Edit /etc/asterisk/logger.conf to include: debug = notice,warning,error,debug,verbose Then start Asterisk using: /usr/sbin/asterisk -vv -g -dd -c /QUOTE /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] DTMF Caller ID w/o polarity inversion
Renato Mintz wrote: Hi Folks, I've been looking around and found some references of some Caller ID patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden and UK. It's been quite hard to understand what has finally been incorporated to the distribution (if anything) or which patches must be applied in witch snapshot of the repository. I've tried some different approaches but nothing worked and my question finally is: Is there any implementation for X100P or TDM400 that supports DTMF caller ID WITHOUT the need of polarity inversion before the DTMF spill? Is anyone working on this? This is the way it works in Brazil and some other coutries... Thanks a lot, Renato Renato, Bug id=9 in the bugtracker is not currently inserted into CVS but can be applied to it. The additional patch for X100P that I put there (srathje) is a quick (ugly) hack for the Danish CID system that uses the DTMF decoder made by egnarf. In Denmark we have no warning (officially) before the CID is received but by monitoring the line I found that a short burst or signal is received just before the CID so I modified the wcfxo.c code to look for this. There is a different approach in bug=1719 for monitoring UK BT CID as it seems to be the same problem, no warning before the CID (V23 FSK) is received. The method used here is a history buffer used to capture the CID and decode when the first ring is detected. I guess you will have to investigate both solutions and use what you can to make a viable solution. If you are successfull in doing so please share with others via the bugtracker.. :-) Regards Soren Rathje ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] UK Callerid bug #1719 TDM400p
Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: VoIP SPAM, what's next ?
Walt Reed wrote: On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said: On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote: hank [EMAIL PROTECTED] wrote: voip spam? I have never gotten any yet. It's is just waiting for the first one to arrive.. The mechanics are just too appealing for spam-like businesses. Why stop there--you can beam pre-recorded messages to phones without a person or phone line ever being involved. You could send hundreds of calls per minute without paying for more then a cheap PC and a T-1. Well, this seems like an DNSBL would be helpful. DNSBL's work quite well at shutting off the big spammer networks. VoIP spam from broadband / DSL / dialup shouldn't be as bad due to the limited bandwidth unless they do the backdoor / trojan / viruses like email spammers do. A VoIP based Beagle e-mail worm with it's own SIP stack ??? LOL... Hmm.. Maybe I should air this on grc.com, some of the propellarheads there may have a clue... /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: VoIP SPAM, what's next ?
John Todd wrote: At 10:09 PM +0200 on 8/10/04, Soren Rathje wrote: John Todd wrote: At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote: Gang, [snip] /Soren It is the mark of an educated mind to be able to entertain a thought without accepting it. - Aristotle Ok, so we moved here from *-dev, no problem... ;-) VOIP Spam is actually pretty trivial to take care of, if only the manufacturers would wise up. We're in the same place we were with SMTP about twelve years ago. I'm sure we'll see a slew of patents and chest-pounding by people with obvious or trivial solutions - welcome to the New WIPO World. The solution is simple: End devices should have the option to only accept authenticated requests. If IP Telephony is supposed to grow up/mature into a technology that will replace TDM over time, this is not an option unless you are building whitelists of gigantic proportions... You're jumping to an overly broad conclusion that whitelists must be created to contain the whole world. I'm saying that I don't know the combination that will best serve the purpose: whitelists, blacklists, chains of trust, PKI, whatever. I _do_ know that if my phone keeps accepting calls from everyone without using _any_ method of authentication, that we can't even debate what method is best or worst, because the calls will just pour in through the unprotected last device in the path. That's pretty simple, but that is the key to the whole solution. However, most end devices will blindly accept any call that they're given, so long as the destination number is correct. I've seen a few phones (Polycom is the only one that comes to mind) which will challenge INVITEs. SIP devices are pretty smart, but I don't think they're capable of being totally smart. The proxy in the middle will have to retain some intelligence and reference some type of permissions model or database to allow calls through or not. I trust that industry (and quasi-industry, like Asterisk) programmers will come up with dozens of ways of intercepting and thrashing unsolicited phone call, so long as there is no back door that the spammer can sleaze through to get right to the desktop. It challenges the concept of e164.arpa. Eh.. The above line was supposed to go in a reply to someone else.. *embarrased* What I meant to say (write) was that I agree with you but that would mean that P2P VoIP is no longer an option unless the same logic is introduced into UA's. Oh shute, the reply went to a completely different list... I do not agree with that at all. You're putting words into the mouths of the users of the service. Not everyone wants their phone on the open Internet; most people don't, actually (regardless of opinions on this list.) If the do not call database of standard telephony in the USA is any reflection of typical opinion on how telephony should be regulated from unrestricted inbound calling, I would say that most people will be overjoyed to have their calls filtered by an intelligent proxy in the future, so long as they have control of the process. I suspect that my SIP device will not have much say in this future world of authentication, other than taking it's marching orders only from authenticated sources, and perhaps having some features to allow me to (during/post call DTMF, or screen-based, or voice commands, or whatever) add a caller to a whitelist or blacklist. I guess that callers originating from China (an example) do not really care about the american DNC database nor do they have access to it. VoIP have no borders between countries or carriers. I can call you from Denmark and have a direct connection, would you require me to do a lookup in the US DNC before calling you?? I have no law here that says I must do that. Sure it would be a polite thing to do, but if I were a spammer I couldn't care less. This says nothing about the option of keeping your phone unprotected and accepting any call. e164.arpa doesn't make any judgement on what the endpoint is of a call, be it a proxy or a UA or something else, nor does it speak to the acceptance of a call - it merely says if you want to reach this number, send the call to this IP address for further instructions. TLS SIP is also a nice concept, since it would require some sort of root authentication that could be revoked or at least recognized if a spam origin was adequately recognized. This is all starting to sound a lot like an anti-spam thread, so I'll stop here. Most intelligent people on the list should be able to figure out a bunch of ways to prevent spam, but the primary one is accountability of origin. Anything that allows that accountability to be compromised from the perspective of the destination means that spam will inevitably slide in, so it is our job to enforce sane authentication/authorization mechanisms NOW on the vendors from whom we
Re: [Asterisk-Users] X100P outbound only (Don't answer)
David Cook wrote:
I tried implementing my * and it didn't pass the spouse factor at this
time. I wanted to hook it up for outbound only at this point to get a
better handle on the dial plans and the echo problem.
I thought this might have been done before as a natural part of
testing - but maybe not.
In wcfxo.c I found this:
if (!wc-offhook !wc-ringdebounce) {
if (!wc-ring (wc-pegcount PEGCOUNT)) {
/* It's ringing */
if (debug)
printk(RING!\n);
zt_hooksig(wc-chan, ZT_RXSIG_RING);
wc-ring = 1;
}
if (wc-ring !wc-pegcount) {
/* No more ring */
if (debug)
printk(NO RING!\n);
zt_hooksig(wc-chan, ZT_RXSIG_OFFHOOK);
wc-ring = 0;
}
}
Is changing the wc-ring = 1 to 0 an appropriate place to fix this
for outbound-only operation?
dbc.
In zapata.conf:
..
signalling=fxs_ks
context=spouse-factor
channel = 1
..
In extensions.conf:
..
[spouse-factor]
exten = s,1,NoOp(${CALLERID})
[next-context]
..
No Wait() or Answer() so the line will never be answered but incoming callerid will be
in the log/cdr... :-)
/Soren
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Re: [Asterisk-Users] CallerID Debug On Zap/POTS Channel
Christopher L. Wade wrote: Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11 11:55:35 ERROR[360464]: callerid.c:193 callerid_feed: fsk_serie made mylen 0 (-26) Aug 11 11:55:35 WARNING[360464]: chan_zap.c:5126 ss_thread: CallerID feed failed: Success Aug 11 11:55:35 WARNING[360464]: chan_zap.c:5168 ss_thread: CallerID returned with error on channel 'Zap/4-1' Or something very similar. Does anybody have any idea how to debug this so I can find out what needs to happen for my CallerID to work. I have changed almost every single setting in this file. My zapata.conf is: [channels] usecallerid=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 context=internal group=1 immediate=no signalling=fxo_ks channel = 1 context=external group=2 callerid=asreceived immediate=yes signalling=fxs_ks channel = 4 In my extensions.conf is simply added a noop statement to print the callerid for me, but nothing ever shows up. Help? Try immediate=no in context=external. FSK type callerid is usually received after the first ring and before the second ring. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: X100P outbound only (Don't answer)
David Cook wrote: Quoting From: Soren Rathje [EMAIL PROTECTED] No Wait() or Answer() so the line will never be answered but incoming = callerid will be in the log/cdr... :-) /Soren I think I just missed something very fundamental. You are saying that the switch doesn't pickup the PSTN line until one of the choosen destinations performs an action like answer/dial, etc? I thought the switch picked up first, then routed the call based on the dial plan. So I can set usedistinctivering=yes with only an answer disposition/context on dring2 causing * to only pickup if you call that number!!! Very cool. This example will answer the call and route to extension.. exten = s,1,Answer() ; RING RING - CLICK exten = s,2,Playback(transfer) ; Please hold while I transfer your call exten = s,2,Dial(SIP/100,20,m) ; 20 sec. with MOH and then give up. This example will route the call and wait for the extension to answer the call... exten = s,1,Dial(SIP/100) ; RING RING RING .. On the other hand, if you specify.. exten = s/5551234567,1,Dial(SIP/100) Your SIP phone will only ring if incoming CID is 5551234567. I have never tried distinctive ring as I do not subscribe to the service but Google would know.. :-) http://www.voip-info.org/wiki-Asterisk+config+zapata.conf dring1context, dring2context, dring3context: Along with setting up to three distinctive ring patterns with dring1, dring2 and dring3, you also set corresponding contexts for incoming calls matching those distinctive ring patterns to jump into. If an incoming call does not match any of the distinctive ring patterns defined, then of course it will enter Asterisk with the default context defined for this channel. dring1context=line2incoming dring2context=business dring3context=chocolate /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: VoIP SPAM, what's next ?
John Todd wrote: At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote: Gang, [snip] /Soren It is the mark of an educated mind to be able to entertain a thought without accepting it. - Aristotle Ok, so we moved here from *-dev, no problem... ;-) VOIP Spam is actually pretty trivial to take care of, if only the manufacturers would wise up. We're in the same place we were with SMTP about twelve years ago. I'm sure we'll see a slew of patents and chest-pounding by people with obvious or trivial solutions - welcome to the New WIPO World. The solution is simple: End devices should have the option to only accept authenticated requests. If IP Telephony is supposed to grow up/mature into a technology that will replace TDM over time, this is not an option unless you are building whitelists of gigantic proportions... That's pretty simple, but that is the key to the whole solution. However, most end devices will blindly accept any call that they're given, so long as the destination number is correct. I've seen a few phones (Polycom is the only one that comes to mind) which will challenge INVITEs. SIP devices are pretty smart, but I don't think they're capable of being totally smart. The proxy in the middle will have to retain some intelligence and reference some type of permissions model or database to allow calls through or not. I trust that industry (and quasi-industry, like Asterisk) programmers will come up with dozens of ways of intercepting and thrashing unsolicited phone call, so long as there is no back door that the spammer can sleaze through to get right to the desktop. It challenges the concept of e164.arpa. TLS SIP is also a nice concept, since it would require some sort of root authentication that could be revoked or at least recognized if a spam origin was adequately recognized. This is all starting to sound a lot like an anti-spam thread, so I'll stop here. Most intelligent people on the list should be able to figure out a bunch of ways to prevent spam, but the primary one is accountability of origin. Anything that allows that accountability to be compromised from the perspective of the destination means that spam will inevitably slide in, so it is our job to enforce sane authentication/authorization mechanisms NOW on the vendors from whom we buy equipment/firmware. Right, the sole purpose of the original post (in asterisk-dev) was to figure out how aware people are of this potential problem and also if people think of this as a problem. /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] personal voicemail
Altus Snyman wrote: At the moment,if you don't get a answer for 10 seconds it goes to voicemail saying something likethe user ate extension 101 is unavailable please leave a message... [snip] http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20VoiceMailMain 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Change your password * Return to the main menu /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 doesn't see remote hangup ? euro-isdn
Walter Klomp wrote: Hi [snip] However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... [snip] If memory serves me well (moved back to DK a year ago) then this is normal Singtel behaviour for subscriber-to-subscriber calling (it's so you can hang-up and go to another room, pick-up and continue). How long time before you see a hangup if you leave the PSTN side on-hook after the call ?? -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. Eh... Sort of like shadow lines ??? Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even activate/deactivate this from the Manager Station Why make it harder than it really is ??? I believe this is exactly what they do when programming your regular (old world) PBX systems... -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New Asterisk bounty: SIP simultaneous registry
Daniel Jimenez wrote: Soren Rathje wrote: Eh... Sort of like shadow lines ??? Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even activate/deactivate this from the Manager Station Why make it harder than it really is ??? I believe this is exactly what they do when programming your regular (old world) PBX systems... -- Soren Explain to me how the AA user knows who's line is ringing? Hmm? To them, it's no different then a call directly to them. The bounty stands. Hmm, Googling around a bit I think I understand what you are looking for... Asterisk is missing functionality similar to Cisco CallManagers m option on ephone configuration. The m option allows you to monitor a specific DN but you do not own it... Right ?? If so, Asterisk will have to handle two types of registers. One To: per DN and any number of Cc:'s... Question is if SIP supports this and as I read the RFC it does not. I wonder if this is built into the Asterisk SCCP stuff.. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 differentiate a *busy* call from not available?
Jean-Yves Avenard wrote: There's just what thing I can't figure out. What is the action for s-. It's the better safe than sorry option... :-) Basically it's a wildcard option, anything beginning with s- will go there... -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] internal external SIP
From: Jon Lawrence Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. .. snip .. to 82.145.37.29:5060 Jul 9 12:41:49 WARNING[5126]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 7712 (Response) set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 81.168.4.69, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 81.168.4.67:5060;branch=z9hG4bK201b0b71 From: 2003 sip:[EMAIL PROTECTED];tag=as3f8ccbff To: sip:[EMAIL PROTECTED];tag=0939785f3bc7641e Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Ok, then I suggest you have a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 differentiate a *busy* call from not available?
Jean-Yves Avenard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
I'm trying to find a way to differentiate wether a SIP extension is
currently busy (e.g. on the phone) or not registered.
So i do something like:
exten = 100,1,Dial(SIP/foo,20,tr)
exten = 100,2,VoiceMail,u100
exten = 100,102,VoiceMail,b100
If the phone doesn't answer I get the message: User is not available
if the phone is currently in used i get the message: User is on the
phone
But if the phone is unplugged, I also get the message: User is on the
phone!
Based on extensions.conf.sample from CVS-HEAD...
[macro-stdexten]
;
; ${ARG1} - Extension
; ${ARG2} - Device(s) to ring
;
; ${DIALSTATUS} = CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL
exten = s,1,Dial(${ARG2},20,tmg)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CHANUNAVAIL,1,Playback(tt-monkeysintro)
exten = s-CHANUNAVAIL,2,Voicemail(u${ARG1})
exten = s-CHANUNAVAIL,3,Goto(default,s,1)
exten = s-CONGESTION,1,Playback(tt-somethingwrong)
exten = s-CONGESTION,2,Voicemail(u${ARG1})
exten = s-CONGESTION,3,Goto(default,s,1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-ANSWER,1,Playback(demo-thanks)
exten = s-ANSWER,2,Hangup()
exten = s-CANCEL,1,Playback(demo-thanks)
exten = s-CANCEL,2,Hangup()
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
-- Soren
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Re: [Asterisk-Users] internal external SIP
Hi all, I've got a problem with external sip clients. My * box has 2 nics, one to my internal network and one on a public IP. There are external sip clients (on public IPs) and internal clients on the internal nic. both clients can register fine. I can phone external clients from the internal clients and the connection works perfectly. But if an external client phones an internal one, the internal phone rings, but when the phone is picked up the external call disappears. Both internal and external have canreinvite=no Can anyone give me any ideas where to start looking into this. bindaddr = 0.0.0.0 ; Local interface externip = xxx.xxx.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks localnet = 10.0.0.0/255.0.0.0; Also RFC1918 localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really without a trace.. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Remote SIP client HACK JOB
This is what keeps my (CVS-HEAD) server happy.. bindaddr = 192.168.0.200 ; Local interface externip = 80.63.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.255.0 ; Local LAN, internal clients etc. (localnet can be repeated for each local LAN segment) Server is firewalled and UDP 5060 and 1-2 is forwarded to internal address. External SIP clients have nat=yes, I dont see any difference with local clients if they have nat=yes or nat=no. -- Soren - Original Message - From: Ryan Courtnage [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 6:31 PM Subject: [Asterisk-Users] Remote SIP client HACK JOB I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: - Asterisk's firewall forwards 5060 udp and 1-2 udp to * - The SIP client's firewall forwards 5060 udp and 1-2 udp to the SIP client - sip.conf contains NAT=YES for this particular client - The SIP client has no special settings, just the external IP of Asterisk's firewall for the SIP Server. I couldn't be happier with the simplicity of this ... however, here is the HACK JOB I need to perform to get the external SIP client's audio to work: When I first start up Asterisk, I need the following In SIP.CONF's [genera] section: - bindaddr = 0.0.0.0 This allows all my internal office phones to work, and also allows me to dial to/from my external client. However, the external client will hear/send no audio. To allow the external client to hear/send audio, I have to change sip.conf ... - bindaddr = EXTERNAL IP ... followed by issuing a RELOAD at the * CLI. It's a total hack, cause if I try to START/RESTART Asterisk with bindaddr=EXTERNAL IP, neither the internal or external clients will work, and I'll just see a bunch of this on the console: Jun 30 15:51:28 WARNING[-1275102288]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x80f680c (len 465) to 66.18.203.117 returned -1: Bad file descriptor My question is ... Is there a better way to do this, without the use of STUN or proxy servers? Thanks -- .. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H323 - IAX
Martin,
Did you change the context parameter in oh323.conf and if you did, did you change it
to an appropriate context in your extensions.conf where musimi.dk can be called ??
The reason I ask is that I've just spend 1 1/2 hour (it was only a matter of time
before I had to, so...) compiling and installing PWLib 1.6.6, openh323 v1.13.5 and
oh323 0.6.3a onto my Fedore Core 1 with Asterisk CVS-HEAD-07/02/04-13:46:17.
Mind you, I followed the README from asterisk-oh323-0.6.3a.tgz to the letter...
I copied my sip-incoming context to h323-incoming in extensions.conf, changed
context=voip-h323 to context=h323-incoming and inBandDTMF=yes in oh323.conf.
Fired up NetMeeting (Yeah.. I'm a Windows user.. So what..) and made a call to a
friend on FWD via IAX2.
The copy/pate stuff below is from a test context I made smilar to your trace...
*CLI oh323 show conf
Configuration of OpenH323 channel driver
Version: 0.6.3
Listening on address: 192.168.0.200:1720
Gatekeeper used: No Gatekeeper
FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF
Supported formats in pref. order: ALAW0
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 4
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 10
-- Executing NoOp(OH323/R2054, MSNetmeeting - 06000) in new stack
-- Executing Dial(OH323/R2054, IAX2/demo:[EMAIL PROTECTED]/6000) in new stack
-- Called demo:[EMAIL PROTECTED]/6000
-- Call accepted by 212.130.58.212 (format ALAW)
-- Format for call is ALAW
-- IAX2[212.130.58.212:4569]/1 answered OH323/R2054
-- Hungup 'IAX2[212.130.58.212:4569]/1'
== Spawn extension (local-h323-inbound, 06000, 2) exited non-zero on 'OH323/R2054'
-- Hungup 'OH323/R2054'
-- H.323 call 'ip$192.168.0.65:3801/2054' cleared, reason 4 (Cleared by remote
user)
I think you need to revise your dialplan and incoming context, something like this for
starters...
-- extract from extensions.conf --
[voip-h323]
;
; OH323 default context from oh323.conf
; Dial 0[number]
;
exten = _0.,1,NoOp,${CALLERID} - ${EXTEN}
exten = _0.,2,Dial(IAX2/demo:[EMAIL PROTECTED]/${EXTEN:1})
exten = _0.,3,Hangup()
-- Soren
- Original Message -
From: Martin Kiefer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 02, 2004 11:48 PM
Subject: [Asterisk-Users] H323 - IAX
Hi there
I am pretty close on giving up on Asterisk :-/
I am (still) trying to make a call from a H323 phone to an Asterisk
provider using AIX. But H323 does not route the number to AIX. All it is
transmitting is an s.
*CLI -- Executing Dial(OH323/R27865,
IAX2/demo:[EMAIL PROTECTED]/s) in new stack
-- Called demo:[EMAIL PROTECTED]/s
Jul 2 23:43:55 WARNING[-1137550416]: chan_iax2.c:5231 socket_read: Call
rejected by 212.130.58.212: No such context/extension
-- Hungup 'IAX2[demo]/3'
== No one is available to answer at this time
The dialed should have been 6000 both it doesn't... Anyone knows why
not?
I have installed the asterisk from cvs using openh323_1.13.5 and
asterisk-oh323-0.6.3a.
I have placed this line in my extensions.conf:
TRUNK=IAX2/demo:[EMAIL PROTECTED]
exten = _.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
I am using the default settings in my oh323.conf. Am I missing something
in this file?
Best regards
Martin Kiefer
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Re: [Asterisk-Users] Asterisk on 64bit ?
From: Nicholas Bachmann Kevin Walsh wrote: Dr. Rich Murphey [EMAIL PROTECTED] wrote: How do you balance the number of active connections per server? In theory, you could use a load balancer. That's as long as you can share the SIP/IAX registrations between the nodes. I'm not sure if that can be done yet - I haven't looked into it. It can. SIP registration info can be stored in a database; see http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers Nick Well, you could look into the ast_data project. It's about dynamic configuration of SIP, IAX, Extensions and Voicemail via ODBC, MySQL and PostgreSQL. (from the author) http://lists.digium.com/pipermail/asterisk-dev/2004-June/004538.html The testing I have done so far looks *very* promising and the potential is there to further extend this and make Asterisk function similar to a neural network. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Serious issues with current CVS?
Damn... I just upgraded my Fedora Core 1 from .2188 to .2194, I thought that was the problem so I backtracked to .2188 only to find my kernel-source garbled so I'm downloading it again... I could have saved myself the trouble.. :-) Oh well... Something good came out of it, I learned how to backstep from one kernel release to another without crashing everything.. -- Soren - Original Message - From: TC [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 3:18 AM Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? YUP lots of total weridness i am trying to track down 1) on sip devices no DTMF, and endless msg reting to get a lock 2) ast_masq deadlocking ..stay tuned but stuff is seriously broken or I am on best drugs i have had in years - Original Message - From: Wojciech Tryc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 23, 2004 6:01 PM Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? Same here, I also lost DTMF on some SIP devices (Grandstream phones) and fax detection on Zap devices. W. Is anybody else having serious issues with the current version from CVS? I just compiled and installed it and: 1) I was able to establish one and only one call before things went weird. 2) It stopped responding to IAX calls after the first. Completely ignored any subsequent commands, including hangup. 3) It stopped responding to CLI commands. 4) The only way to kill it was to use kill. Is anybody else experiencing this? Thanks, Steven Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Unify Incoming and Outgoing sound files
- Original Message - From: Carlos Medina [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 22, 2004 10:47 PM Subject: [Asterisk-Users] Unify Incoming and Outgoing sound files Hi, i have a call center which receives many calls at day. Those calls are stored in a directory in my asterisk server as WAV files. The problem is that each call is divided in 2 files: an IN.WAV file and OUT.WAV file. The OUT.WAV file is what im speaking to other person, the IN.WAV file is what that person is speaking to me. I need to unify that files into one complete file that consolidate a complete call, i dont know if there is an application that can do that. Thanks for your help Carlos Andres Medina. http://www.google.com/search?q=site:www.voip-info.org+merge+recorded+calls should give you: http://www.voip-info.org/wiki-Monitor+setup+sample Half way down the page is what I presume you are looking for. quote Don't forget to make the /var/spool/asterisk/monitor directory! Finally, mix_monitor_files.pl does the mixing job and combines the in and out files /quote -- Søren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Testing UK emergency dialing and LCR.
- Original Message - From: Storer, Darren [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 11:27 AM Subject: RE: [Asterisk-Users] Testing UK emergency dialing and LCR. It became apparent, back in 1999, when I was part of a team providing consultancy to a UK Telco for VoIP VPN launch, that a POTS line would be required locally at each branch office for power-fail compliance and to ensure that the OACs (Operator Assistance Centres) did not get confused about which location the emergency call was originated from. We discussed spoofing the branch office CLI in network at an SS7 level but that idea was shelved as there would have to have been an associated POTS line entry in the OAC database in the first place. At that time Cisco CPE had no way of utilising the power-fail POTS lines so a red 'phone was provided for use on each floor of the branch offices that only had VoIP VPN telephony. Here in Denmark (where I live) the 112 service also have regular 8 digit phone numbers to be used for this kind of scenarios. Normally your location and associated 112 service centre will be resolved by the SS7 network thus you only need to know about the 112 number, BUT, with the indroduction of our new small IP Telephony companies (5000 customers) they will (manually) map the appropriate 8 digit phone number to your 112 entry when you sign-up bypassing the SS7 logic and dial directly the emergency centre for your area. My bet is that this also applies for other telco's, the question is if they are willing to disclose the information or not. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Status-info 1: Signalling C7 / SS7
- Original Message - From: Roger Schreiter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 16, 2004 7:03 PM Subject: Re: [Asterisk-Users] Status-info 1: Signalling C7 / SS7 Senad Jordanovic schrieb: ... Are you aware of bounties posted to get SS7 working with *? If not look at http://bugs.diuim.com . ... No, I'm not. Unfortunately, the link you mention, does not work. I tried http://bugs.digium.com and found a login form. I logged in with anonymous and looked around, searched some keywords, but did not find anything matching your message. Please tell me more hints about what you are writing! Thank you! Roger. Google is your friend... http://www.google.com/search?q=site:www.voip-info.org+asterisk+bounty finds.. http://www.voip-info.org/wiki-Asterisk+bounty+SS7 apparently worth USD 3,000 -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail problem
- Original Message - From: Sean Garland [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 9:53 PM Subject: RE: [Asterisk-Users] Voicemail problem How do you specify sendmail, or any mail program? I changed the servermail= to equal my in-house exchange server, and allowed relaying by it's the pbx's IP address, but I still don't understand how it know where to send or what program it uses.. In voicemail.conf.. fromstring=The Asterisk PBX ; Change the From: string serveremail=asterisk(atsign)domain.com ; Who the e-mail notification should appear to come from mailcmd=/usr/sbin/sendmail -t ; You can override the default program to send e-mail Note: fromstring serveremail are translated into The Asterisk PBX asterisk(atsign)domain.com in your email From: address. section of email header Date: Wed, 09 Jun 2004 21:04:28 +0200 From: The Asterisk PBX asterisk(atsign)domain.com To: Soren soren(atsign)domain.com Subject: New VM (1) - 2:04 long in mailbox 100 from Joe User 12345678 Message-ID: Asterisk-1-100-2792(atsign)asterisk.domain.com On your server you install/setup/configure sendmail and have it point to your normal mailserver as relaymailer, this way you can control who gets what from your normal mailserver. As fas as I remember, the only thing I changed in /etc/var/sendmail.cf was: # Smart relay host (may be null) DSmail.domain.com to enable relaying via mail.domain.com. It may not be the most secure way to do stuff, but I have everything behind NAT and Firewall with SMTP traffic only allowed to/from my regular mailserver. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] phone calls betweens phones behind the same nat
if Asterisk publishes the external IP address in the SIP messages you get trapped in the NAT Trap.. internal IP1 - NAT - external IP - NAT - internal IP2 He.. Take a string/thread/wire, make a loop and put it through the hole in a coin, place a match/stick in the loop going through the coin, pull the string tight so the match/stick is flat against the coin. Now, pull one end of the string while holding the coin Get it ?? Not that many devices can do this... And if the do, they cost really big money... :-) -- Soren - Original Message - From: Didelot Loic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 11, 2004 2:37 PM Subject: [Asterisk-Users] phone calls betweens phones behind the same nat Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] phone calls betweens phones behind the same nat
BTW..
Q1: Are you using HEAD or STABLE?
Q2: Are you using bindaddr, externip, localnet and localmask in your SIP.CONF?
Something tells me that the CVS HEAD version of Asterisk is smart enough to figure out
if we'r inside or outside...
(chan_sip.c checked out june-10)
/*--- ast_sip_ouraddrfor: NAT fix - decide which IP address to use for ASterisk
server? ---*/
/* Only used for outbound registrations */
static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
{
/*
* Using the localaddr structure built up with localnet statements
* apply it to their address to see if we need to substitute our
* externip or can get away with our internal bindaddr
*/
Then again, I could be wrong.. ;-)
-- Soren
- Original Message -
From: Didelot Loic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 11, 2004 2:37 PM
Subject: [Asterisk-Users] phone calls betweens phones behind the same nat
Hi,
I have the following problem.
I have 5 phones behind the same nat (canreinvite=yes). it works fine to
receive calls and to make calls. sound quality is good, so everything
works fine.
The poblem is that the phone behind nat cant call each other. It works
if canreinvite=no. But i want to do this.
Does anyone have an idea?
Regards,
cjk.
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Re: [Asterisk-Users] Psssst. The US is asleep - let's talk intern ationalization !!!
From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 14, 2004 11:41 AM Subject: Re: [Asterisk-Users] Pt. The US is asleep - let's talk intern ationalization !!! Robinson Tim-W10277 wrote: And let's also spell things properly! Like 'internationalisation' ...'Weasels have got into your phone system' instead of 'gotten into your phone system...' And 'please press the hash key..' instead of 'pound key' There should probably be en_uk, en_us, en_ca, en_za, en_nz, en_oz, en_ie and en_in etc to allow each English-speaking country to localise prompts. and en_se for my swenglish :-) Could something like this be of any use in the internationalisation process ?? http://www.softintegration.com/products/sdk/embedch/ (commercial license) Embedded Ch allows you to embed Ch (C/C++ interpreter) into your C/C++ application programs and hardware. With Embedded Ch, your C/C++ binary applications in the runtime can call ch scripts/functions, and the ch scripts can call back C/C++ binary functions. You can even modify ch scripts without recompiling your C/C++ applications. It can make your applications more user-friendly and powerful with extension of C/C++ scripts. Anyone know of a similar free product ?? -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] I love you!
Nah.. The only way I ever notice that someone sent me a virus is when my mailserver says beep.. And that is currently so frequent that I'm considering turning that off.. :-) Makes me think of this (The Network Auralizer): http://peep.sourceforge.net/intro.html Check out the Low load average demo mp3, you'll know what I mean.. -- Soren - Original Message - From: Thomas Gallaway [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 9:00 PM Subject: Re: [Asterisk-Users] I love you! [EMAIL PROTECTED] wrote: lovely, :-) Is it just me or where there allready 3 virus sent to this list today? Maybe time for denim to disallow attachments? :-) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] One, två, tre, quatre, cinq ... International numbers in say.c
I did a quick test with the danish numbers in say.c patch (04-20-04 02:11) and found this.. *1 -- Executing SayNumber(SIP/1000-497f, 1) in new stack -- Playing 'digits/1' (language 'da') *2 -- Executing SayNumber(SIP/1000-497f, 100) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/hundred' (language 'da') *3 -- Executing SayNumber(SIP/1000-497f, 101) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/hundred' (language 'da') -- Playing 'digits/1' (language 'da') *4 -- Executing SayNumber(SIP/1000-497f, 1000) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/thousand' (language 'da') -- Playing 'digits/and' (language 'da') -- Executing SayNumber(SIP/1000-497f, 1001) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/thousand' (language 'da') -- Playing 'digits/and' (language 'da') -- Playing 'digits/1' (language 'da') *5 -- Executing SayNumber(SIP/1000-497f, 100) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/million' (language 'da') -- Playing 'digits/and' (language 'da') -- Executing SayNumber(SIP/1000-497f, 101) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/million' (language 'da') -- Playing 'digits/and' (language 'da') -- Playing 'digits/1' (language 'da') *1)pronounced en, not an issue in itself but see next point. *2)pronounced et + hundrede, different digit 1 et. *3)pronounced et + hundrede + og + en, there is an og missing. *4)pronounced et + tusinde, no need for the og *5)pronounced en + million, no need for the og A few pointers to how it is done... (Last time I translated VoiceMail software was 7 years ago and the biggest problem was making our vendor understand that we needed two different 1's. ) (1,2,3...99) en, to, tre...ni|og|halvfems (100,101...199) et|hundrede, et|hundrede|og|en, ..., et|hundrede|og|ni|og|halvfems (1000,1001...1099) et|tusinde, et|tusinde|og|en, ..., et|tusinde|og|ni|og|halvfems (1100,1101...1999) et|tusinde|et|hundrede, et|tusinde|et|hundrede|og|en, ..., et|tusinde|ni|hundrede|ni|og|halvfems (100,101...199) en|million, en|million|og|en, ..., en|million|og|ni|og|halvfems (1000100...199) en|million|et|hundrede, ..., en|million|ni|hundrede|ni|og|halvfems|tusinde|ni|hundrede|ni|og|ni|og|halvfe ms (200...X) to|millioner...X -- Soren - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk [EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:53 PM Subject: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c http://bugs.digium.com/bug_view_page.php?bug_id=0001429 [SNIP] * If we all work on this together quickly, we may have a working say.c in the CVS soon. But to even ask a committer for support, I need test results up there on the bug tracker. * Thank you for your support! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit on call in minuttes.
I saw in the current CVS version that a new parameter has been added to app_dial.c... The option string may contain zero or more of the following characters:\n 't' -- allow the called user transfer the calling user\n 'T' -- to allow the calling user to transfer the call.\n 'r' -- indicate ringing to the calling party, pass no audio until answered.\n 'm' -- provide hold music to the calling party until answered.\n 'H' -- allow caller to hang up by hitting *.\n 'C' -- reset call detail record for this call.\n 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n 'g' -- goes on in context if the destination channel hangs up\n 'A(x)' -- play an announcement to the called party, using x as file\n 'S(x)' -- hangup the call after x seconds AFTER called party picked up\n so that should enable you to do the following: Call timeout = 20 sec Max Call Duration = 300 sec = 5 min. exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300)) however, I have not tried it yet so someone correct me if I am wrong.. -- Søren - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 07, 2004 10:04 AM Subject: RE: [Asterisk-Users] Limit on call in minuttes. exten = 1,AbsoluteTimeout ($SECONDS) Ta SJ Hi, I saw somewhere that it was possible to set a limit for how long time a call could be, for an extension in extension.conf. But I can't find it anymore. Can someone please help. Calls to '411' an operator may max. be 5 min. I have this in extension.conf. [shortcuts] exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) [operator] exten = 0,1,Dial(SIP/operator,30,tr) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Limit on call in minuttes.
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
From extensions.conf:
[pstn-out-nat]
;
ignorepat = 0
; NOT USED
exten = _0XX0X,1,Congestion
; Local eight-digit dialing accessed through trunk interface
exten = _0NXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},20,S(20))
exten = _0NXXX,2,Congestion
From * console:
-- Executing Dial(SIP/1000-4d25, Zap/1/4060|20|S(30)) in new
stack
-- Setting call duration limit to 30 seconds.
-- Called 1/4060
-- Zap/1-1 answered SIP/1000-4d25
-- Hungup 'Zap/1-1'
== Spawn extension (default, 04060, 1) exited non-zero on
'SIP/1000-4d25'
cdr_odbc: Query Successful!
-- Søren
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec = 5 min.
exten =
411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300))
however, I have not tried it yet so someone correct me if I am wrong..
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Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] HMM - This wont work :( exten = 10,1,Dial(SIP/hha1,20,S(10)) exten = 10,2,VoiceMail,u10 exten = 10,102,VoiceMail,b10 When did you checkout your version of Asterisk from CVS ?? This feature was put into CVS on the 6'th as a fix for bug #1107 but I have not seen it in v1-0_stable. -- Søren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP extension busy when not available ??
I use ChanIsAvail() to check to see if the phone is connected at the top of the dialplan for that extension. This works for IAX2 and SIP channels but not for MGCP. If you are interested in the actual code I can send it to you from home tonight. Robert Thank you, yes please... Well, I'm about three weeks into my very first * installation (that sort of works), so basically any info/tips/tricks/word of advice is accepted with appreciation... -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP extension busy when not available ??
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:45 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension.
The Macro is:
[macro-stdexten]
exten = s,1,ChanisAvail(${ARG2})
exten = s,2,Dial(${ARG2},20,Ttr)
exten = s,102,GoTo(voicemail,s,1)--Note A
exten = s,103,Hangup
exten = s,104,GoTo(voicemail,s,1)--Note B
exten = s,105,Hangup
Hey, that works pretty cool...
I've changed it a bit... (the DND stuff I found elsewhere)
[macro-stdexten]
exten = s,1,DBget(temp=DND/${ARG1}); DND set ?
exten = s,2,Goto(104) ; Yes.
exten = s,102,ChanisAvail(${ARG2}) ; Channel up?
exten = s,103,Dial(${ARG2},20,tr) ; Ring the interface, 20 seconds
maximum
exten = s,104,Voicemail(u${ARG1}) ; Send to voicemail w/ unavail
announce
exten = s,105,Hangup ; Doh...
exten = s,203,NoOp ; Nada...
exten = s,204,Voicemail(b${ARG1}) ; Send to voicemail w/ busy announce
exten = s,205,Hangup ; Doh...
[dnd]
; *61# turns it on
; *60# turns it off
;
exten = _*61,1,DBput(DND/${CALLERIDNUM}=YES})
exten = _*61,2,Playback(vm-goodbye)
exten = _*61,3,SoftHangup
exten = _*60,1,DBdel(DND/${CALLERIDNUM})
exten = _*61,2,Playback(vm-goodbye)
exten = _*60,3,SoftHangup
Now I just have to figure out a way to tell if it's on or not..
-- Soren
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Re: [Asterisk-Users] SIP extension busy when not available ??
- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 22, 2004 8:52 PM Subject: Re: [Asterisk-Users] SIP extension busy when not available ?? Although the current logic does not require a sip phone to register, it would seem like the asterisk logic should be something like: a. call is attempted to sip x1234, b. if * knows the extn is in use, return busy, or, c. if not busy, asterisk attempts to contact x1234 across the wire, d. if no contact, return Unavailable Or if not registred. In a wireline telephony scenario the above would be the proper method as we do not know if the subscriber have their phone plugged in or not. With Asterisk we experience the same information as mobile operators do: unreachable, unanswered and busy. IMHO we should have the same options. So, the priority for a type=friend would be: a: check if client is registered and/or reachable, if not - return unreachable b: check if client is busy, if call-waiting not active - return busy c: if call is rejected by client, return approriate message d: if call is unanswered, return unavailable or busy with reference to (b). -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP extension busy when not available ??
How come * says 1010 is BUSY in the trace below? I would have guessed
UNAVAILABLE since 1010 is not logged on/registered.
*
[EMAIL PROTECTED] asterisk]# asterisk -fcg
...
Asterisk CVS-02/21/04-15:10:20, Copyright (C) 1999-2001 Linux Support
Services, Inc.
...
...
-- Executing Macro(SIP/1000-2806, stdexten|1010|SIP/1010) in new
stack
-- Executing Dial(SIP/1000-2806, SIP/1010|20|tr) in new stack
Feb 22 02:21:19 NOTICE[278545]: app_dial.c:527 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy at this time
-- Executing VoiceMail(SIP/1000-2806, b1010) in new stack
...
...
sip show peers
Name/usernameHost Mask Port Status
2000 (Unspecified) (D) 255.255.255.255 0UNKNOWN
1020 (Unspecified) (D) 255.255.255.255 0UNKNOWN
1010 (Unspecified) (D) 255.255.255.255 0UNKNOWN
1000/1000192.168.0.65(D) 255.255.255.255 5061 OK (9 ms)
[macro-stdexten];
exten = s,1,Dial(${ARG2},20,tr); Ring the interface, 20
seconds maximum
exten = s,2,Voicemail(u${ARG1}); If unavailable, send to
voicemail w/ unavail announce
exten = s,3,Goto(default,s,1) ; If they press #, return to
start
exten = s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail
w/ busy announce
exten = s,103,Goto(default,s,1); If they press #, return to
start
*
/Soren
--
It is the mark of an educated mind to be able to entertain a thought
without accepting it.
- Aristotle
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[Asterisk-Users] CallerID or Noise ?
As a new member and with less than 2 weeks experience with Asterisk, I allow myself to ask potential dumb questions... I am using a TDM400P and a X100P card in a Celeron 2,6 GHz box with 256 MB memory running RH 8.0 run level 3. In Denmark we use DTMF style CallerID sent between 1'st and 2'nd ring. Now, I've removed/diabled all CalledID settings and enabled immediate answer and this is what I get when my system answers the call *without* the mandatory s,Wait,2: -- Starting simple switch on 'Zap/1-1' -- Sent into invalid extension 's' in context 'default' on Zap/1-1 -- Executing Playback(Zap/1-1, transfer) in new stack -- Playing 'transfer' (language 'en') The question is: How do I enable any kind of debug or whatever to see the information Asterisk don't like (could it be the CallerID data) and can I capture this information and use it ? The format of the CallerID is described in ETS 300 659-1 and ETS 300 659-2 if anyone is interested, and can be found here: http://www.secret100.nm.ru/ets300659.pdf and http://www.secret100.nm.ru/ets3006590e02.pdf (I know, weird links, Google found them for me) Thanks Soren -- It is the mark of an educated mind to be able to entertain a thought without accepting it. - Aristotle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
