Re: [Asterisk-Users] Low Volume WAV Files in Email Attachments
Third, those complaining of low volume in emailed files are usually using a compressed format. In the uncompressed wav format, the volume is effectively doubled by shifting the audio data to the left one bit. This is done at the format level. Of course on playback via asterisk, it checks to see if it needs to shift the audio down and does so. So playback between asterisk recorded wav files should all sound the same on asterisk but isn't the same when played via a normal audio app. The complaints come mainly regarding the emailed attachements, which are WAV49 (MS-GSM) files, which (as far as I can tell) are just justified right and packed into 65 bytes per the IETF I-D. These files are not played back within Asterisk, and honestly, most of what you said above here is rubbish. I just spent more time than I ever cared to spend (including studying the actual GSM codec spec from the ETSI), learning more than I ever cared to learn about GSM (which, btw, if you are concerned about patents, is just as subject to them if you believe Philips' claims), and the difference between uncompressed WAV files (which also suffer from attenuated signal levels) and the GSM and/or MS-GSM files is far far more than just shifting the audio data to the left one bit. There is an issue surrounding the recording of data through Asterisk. That is inarguable. The problem is that no one seems to agree on where to begin looking, so no one has, really. I don't know the origin of the GSM files that make up the Comedian VM system prompts, but they do not suffer from this problem. However, GSM files generated by the VM system, at the least, have a signal attenuation problem to the point that the emailed attachments are unusable, and by most accounts, the phoned-in VM retrieval is barely useful to boot. Not only am I willing to try to track this down, I am furiously taken with the task, because it's a real issue that needs to be addressed, and I do understand that the actual devs have more important things to fix first. That's one of the nice things about open-source, eh? ;) ___ 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] Hardware performance issues - Zaptel / wct4xxp for TE405P
Hi, We have an installation that is expanding rapidly. We are about to move from 1 x E1 towards 4 x E1 (PRI CPE) and therefor we bought a TE405P card. Having set up a test mashine based on a 2,4GHz Celeron CPU and 256MB RAM, all is fine while modprobing a single E1 span (30 B lines + 1 D) on our spare TE100P. BUT: When we add in the TE405P and configure 2 E1 span, the modprobe takes about a half a minute. Optimistically we tested it with all 4 spans available on the card; result was that the mashine never reentered prompt state :-( The kernel used is a Debian 2.4.22 and we generally have Asterisk running on the testbox except from this issue. SMP is disabled and correct processortype selected in kernel. Main gateway is based on a single Xeon 2,4GHz - after this test we worry a bit about performance issues when moving to this platform. Here is some additional data from our set-up (please note that the telco we peer against require that the spans is set up on timeslot level for them to be able to show custumer Caller-ID correctly through their Nokia PRI NET): /etc/zaptel.conf: # E405P PRI span=1,1,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 # global parameters loadzone = nl defaultzone=nl /etc/asterisk/zapata.conf: [trunkgroups] [channels] group = 1 context=default switchtype=national signalling = pri_cpe channel = 1-15,17-31 pri_dialplan=national usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 immediate=no musiconhold=default context=default group = 2 context=default signalling = pri_cpe channel = 43-46, 48-54 callerid=asrecived pri_dialplan=national switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes txgain=0.0 txgain=0.0 immediate=no musiconhold=default Should we run out and buy 2 new 3GHz processors for the server or is the issue here Celeron or even better: are we doing something wrong (cheapest approach :-) Thanks in advance /Niels - Denmark --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.801 / Virus Database: 544 - Release Date: 24-11-2004 ___ 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] Opinions on renice or turning off swap or ramdis k as swap?
Colin Anderson wrote: I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? What asterisk -p does is mark the aterisk process as a POSIX real time priority process. Unless you have other process marked in the same way, the scheduling algorithm will prefer this process to others at all times. which means that if is not blocking, it will be the running process. I've been running like this with Asterisk for a couple of month with no ill effects except that some error conditions cal cause asterisk to go into a loop which will effectively freeze all user space activity on the machine. I keep a shell set to a higher real time priority then asterisk on the machine for these cases. You can use the following tool to get a real time priority shell: http://projects.codefidence.com/realtime.html 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) Don't do that. Swap can be useful to allow the kernel to organize memory a little better and avoid fragmentation. 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? Very bad idea. Linux memory management is much smarter then DOS, from which you got this idea, I assume. Anyone else have any performance tips? Disable any interrupts not needed on the system. Specifically, use NAPI enabled network device drivers and turn NAPI on. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Experiences with Termination Providers?
Me wrote: I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two of them to be offer a quality service with most of the features I want but horrible customer service/support and response times to my questions etc. The other two seem to respond quickly and have great customer service but have awful connections to the web and basically unusable services. Can someone recommend a termination partner for our VOIP Venture that can provide reliable services, good features/DID's and GOOD customer service? Price is important as well but comes last in line after the items mentioned above. As far as I can tell there are no providers that match your requirements. It's the typical growth pattern. Tiny companies have less reliable service, but great customer service. Larger companies have more reliable service, but crappy customer service. If you ever do fine a unicorn, let the rest of us know. --Eric ___ 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] redhat9 100% CPU
TELUX wrote: Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok. I would try running asterisk with LD_ASSUME_KERNEL=2.4.1 if it isn't already. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] Experiences with Termination Providers?
-Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. Well, you must be dreaming :) It all depends on your buying power, if you have at least 2-3 million minutes goto Level3 or broadvox. If you are just starting up and no commitments, then you have to stick with one of the two categories that you mentioned below. I chose to use first type you mentioned. BTW:- if you find a provider which could give those points mentioned and still go with no commitments, please let me know. Cheers Dough -Original Message- From: Me [mailto:[EMAIL PROTECTED] Sent: Saturday, November 27, 2004 11:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Experiences with Termination Providers? I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two of them to be offer a quality service with most of the features I want but horrible customer service/support and response times to my questions etc. The other two seem to respond quickly and have great customer service but have awful connections to the web and basically unusable services. Can someone recommend a termination partner for our VOIP Venture that can provide reliable services, good features/DID's and GOOD customer service? Price is important as well but comes last in line after the items mentioned above. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.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
Re: [Asterisk-Users] Problem with voicemailsystem
[EMAIL PROTECTED] schrieb: After calling the number and no response of our client the voice-box gives response. Thats ok... but after the voice-box, which ist self- configured by our client the server respondes with the notivication to leave your message please speak after... blablabla Does anyone knows a possibility to disable the message of the server and only able the message of our client? Example: client says:Im not in my office, please leave a message. Well, after this message the sever should send the signal and record the opposite, without the message... to leave your message please speak after... blablabla CLI show application VoiceMail -= Info about application 'VoiceMail' =- [..] [Description]: VoiceMail([s|u|[EMAIL PROTECTED][EMAIL PROTECTED]): Leavesvoicemail for a given extension (must be configured in voicemail.conf). If the extension is preceded by * 's' then instructions for leaving the message will be skipped. hth rgds pos ___ 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
Fwd: Re: [Asterisk-Users] very newbie question
On Sat, 27 Nov 2004 19:37:54 +, Corvin [EMAIL PROTECTED] wrote: I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? This is how I do it - Thank you very much to all of you. I have one more question which troubles me. We have scenario: (only SIP is considered now) Subscriber A registered in Asterisk company1.org (eg. [EMAIL PROTECTED]) Subscriber B registred in Asterisk company2.org ([EMAIL PROTECTED]) How it is possible to make connection between those. Their extensions (called party) (maybe I don't understand term extension correctly) is not definied in corresponding extensions.conf. Extension B is not definied in extension.conf in Asterisk A. etc. Again many thanks for any help. :) Regards, Corvin ps. sorry if I douuble my post --- ___ 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] very newbie question
Soemthing goes wrong with this mail list: I am getitng something like it: Sorry. Your message could not be delivered to: Aster risk (Mailbox or Conference is full.) ?? This is rest of my post. On Sat, 27 Nov 2004 19:37:54 +, Corvin [EMAIL PROTECTED] wrote: I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? This is how I do it - Thank you very much to all of you. I have one more question which troubles me. We have scenario: (only SIP is considered now) Subscriber A registered in Asterisk company1.org (eg. [EMAIL PROTECTED]) Subscriber B registred in Asterisk company2.org ([EMAIL PROTECTED]) How it is possible to make connection between those. Their extensions (called party) (maybe I don't understand term extension correctly) is not definied in corresponding extensions.conf. Extension B is not definied in extension.conf in Asterisk A. etc. Again many thanks for any help. :) Regards, Corvin ps. sorry if I douuble my post --- --- ___ 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] SetVar ALERT_INFO
Hello,
I've got my dialplan configured to do a double ring when a customer
service call comes in, and a normal ring when an extension is dialed
directly. When a customer service call is transferred, I want to ring
to revert back to normal.
In the local extension macro, I have the following
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=)
exten = s,n,SetVar(_ALERT_INFO=)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
When a call is transferred, here's what I see on the console
-- Executing NoOp(Zap/1-1, Bellcore-r3) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, ALERT_INFO=) in new stack
-- Executing SetVar(Zap/1-1, _ALERT_INFO=) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
It appears as though both ALERT_INFO and _ALERT_INFO are not set. This
isn't the case, however, because the party receiving the transferred
call still hears the double ring.
Now, I'm not sure if this is design or bug. I think I saw mention of
the _ doing something special but I'm not sure where and see no mention
of it in the wiki under SetVar.
Pointers would be appreciated.
Thanks,
Trevor Peirce
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[Asterisk-Users] asterisk compile errors - pbx_dundi.c -help
Hi, Reviewing the archives I saw /2004-October/070314.html from Tim Lewis. His error is almost identical to mine i.e. when make clean; make install in asterisk sub dir, I get the following: pbx_dundi.c:54:18: zlib.h: No such file or directory pbx_dundi.c: In function `update_key': pbx_dundi.c:1315: warning: implicit declaration of function `crc32' pbx_dundi.c: In function `dundi_decrypt': pbx_dundi.c:1371: warning: implicit declaration of function `uncompress' pbx_dundi.c:1371: error: `Z_OK' undeclared (first use in this function) pbx_dundi.c:1371: error: (Each undeclared identifier is reported only once pbx_dundi.c:1371: error: for each function it appears in.) pbx_dundi.c: In function `dundi_encrypt': pbx_dundi.c:1396: warning: implicit declaration of function `compress' pbx_dundi.c:1397: error: `Z_OK' undeclared (first use in this function) make[1]: *** [pbx_dundi.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make: *** [subdirs] Error 1 What does this error mean and can anyone help me?? Thanks Conor McCleane (Dell Dimension P4 2.8 GHz-HT running SuSE 9.1 Pro) _ Sign up for eircom broadband now and get a free two month trial.* Phone 1850 73 00 73 or visit http://home.eircom.net/broadbandoffer ___ 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: Top posting - are we there yet?
On Tuesday 16 November 2004 17:12, Jay Milk wrote: I'm a fairly reasonable person, and I have yet to see one good argument (and quoting netiquette is not on argument, that's opinion) for bottom-posting. To me, it is terribly inefficient and wastes time, especially when you hide your post between the original message and some ludicrously elaborate signature. Top-posting, to me, is more logical, as it presents the answer in a prominent position. And inline-posting makes sense when you're responding to multiple questions or points in an email... Whether you top post or not is irrelevant really. Top posting - you have to scroll around to find out what question they are answering. bottom posting - you have to scroll to find the answer. I'll reply to both top and bottom postings - if I think I've got anything to add. What's more annoying is people who just click reply instead of starting a new tread. Jon ___ 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] Real time queue monitoring
Hello list, I am glad to announce that XC-AST 0.5, released today, offers real time queue monitoring facilities that let you see the calls flowing through a set of Asterisk queue(s) and agents logging on and off. This way, XC-AST provides an one-stop solution to generate reports, monitor queues and let agents see their own calls and launch external CRM/tracking apps. The software is available for trial, together with an expanded user manual, at http://demo.xcept.it/xc-ast I am also glad to announce that we plan to offer a free licence of XC-AST to smaller call centers, like SOHOs and home PBX. Details are not out yet, but I believe that the average * hacker will be able to use a free XC-AST installation, while bigger commercial call centers will need a proper licence. Please let us know of any bugs or ideas you should encounter. l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] SNOM telephones and LEDs
On 25 Nov 2004, at 21:09, Asterisk wrote: I've just got a Snom 190 phone with which I'm really pleased. I can get the LEDs on the keys to light in response to an extension being in use which is cool, but there's a feature I'd like to implement. i'm very interested in this option, have a snom 190 too. What option do you choose on function keys options menu? Adrià Vidal ___ 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] asterisk based bbs
hello list, I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about asterisk, but a vocal bbs that runs on asterisk, so that people can call, hear the discussion of the day, leave messages, etc. it seems a rather basic application to me though I cannot find much about. thanks l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] am i baned or something?
Soemthing goes wrong with this mail list: I am getitng something like it: Sorry. Your message could not be delivered to: Aster risk (Mailbox or Conference is full.) ?? Regards, Corvin ___ 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] am i baned or something?
Soemthing goes wrong with this mail list: I am getitng something like it: Sorry. Your message could not be delivered to: Aster risk (Mailbox or Conference is full.) ?? Probably nothing to do with you. A lot of people run mail software that isn't fit for consumption even by pigs (or goats or whatver you prefer). In particular, people like to run autoresponders and the like which completely ignore the envelope sender (which is where all backchannel communications, such as errors, ought to go) and instead target the listed From: address in the body of the message, which doesn't necessarily have anything to do with the transaction, other than perhaps having originally authored the message at some past point. This is, of course, generally the fault of the software they run. I'll further note that the software in question is frequently MICROSOFT BROKENWARE. Sadly, the small portion of the Internet community that has a clue does not seem to care enough to do something to deal with this problem, such as finding ways to deliberately cause these mechanisms to break horribly until they're removed or turned off by their clueless MCSE admins. Regards, ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] am i baned or something?
On Sun, 28 Nov 2004 15:37:44 +, Corvin [EMAIL PROTECTED] wrote: Soemthing goes wrong with this mail list: I am getitng something like it: Sorry. Your message could not be delivered to: Aster risk (Mailbox or Conference is full.) This is a problem on your end. I replied yesterday and got a reply back from you with that message. Your messages have been getting through as I have seen you post this 2-3 times now. Leif Madsen. ___ 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] am i baned or something?
In particular, people like to run autoresponders and the like which completely ignore the envelope sender (which is where all backchannel communications, such as errors, ought to go) and instead target the listed From: address in the body of the message, which doesn't necessarily have anything to do with the transaction, other than perhaps having originally authored the message at some past point. This is, of course, generally the fault of the software they run. I'll further note that the software in question is frequently MICROSOFT BROKENWARE. Sadly, the small portion of the Internet community that has a clue does not seem to care enough to do something to deal with this problem, such as finding ways to deliberately cause these mechanisms to break horribly until they're removed or turned off by their clueless MCSE admins. Thank you very much for answer, I was sure that my emails were somehow dropped. I am lucky so that I am not forced to use M$ software. OT. Is there very simple, comprehesive step-by-step explanation, forum for asterisks? I've to introduce to this software very quicly and there some concepts that I don't understand. I've read a lot of web pages, viki's etc. But I can't find any realy good documentation site. Those sentences in handbook like this is document is inteded for geeks or similar causes that all idea is not looking very seriously. But I've to introduce very quicly. So I am reading whatever I find about it. Could you reccomend me something better than handbook, and vo-ip wiki? Regards, Corvin ___ 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] am i baned or something?
On Sun, 28 Nov 2004 16:25:15 +, Corvin [EMAIL PROTECTED] wrote: Could you reccomend me something better than handbook, and vo-ip wiki? If you haven't seen it already, you can also try http://www.asteriskdocs.org. Leif Madsen. http://www.asteriskdocs.org ___ 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] Experiences with Termination Providers?
I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. It is an appropriate question - but I think the 'Welcome to the mailing list' message should point out that this is not a USA only list - anyone who posts this type of message should really say where they want service to and from! Linus ___ 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] asterisk compile errors - pbx_dundi.c -help
On Sun, 28 Nov 2004 11:23:01 +, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: pbx_dundi.c:54:18: zlib.h: No such file or directory I'm going to make an educated guess that you don't have zlib and zlib-development packages installed. These are now required to be installed for the CVS branch. ___ 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] Low Volume WAV Files in Email Attachments
Steven Critchfield wrote: Second, read the rants on licensing. Unless you find a BSD licensed mp3 encoding library and convince Mark of it's need, it is unlikely to make it to the core code base. When snackAmp blew up on GSM-encoded wav files I did some cursory research and found FLAC: http://flac.sourceforge.net/ The license for the libraries is a BSD-variant. I'm not an expert on audio formats so no flames please. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.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
Re: [Asterisk-Users] asterisk based bbs
lenz schrieb: I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about asterisk, but a vocal bbs that runs on asterisk, so that people can call, hear the discussion of the day, leave messages, etc. It doesn't really make sense to me. It only makes sense for some very limited fields. e.g. when somebody cannot read or write (he hasn't learned it or is blind). Everybody else could use the computer. Most people who would use such a system do have a computer at home I guess. 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] [Fwd: Call Transfer between phones]
Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call Thanks ___ 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] [Fwd: Call Transfer between phones]
Jeremy SALMON wrote: I have an PSTN line (X100P) and 10 grandstream budge tone phone. Jeremy, Receive call, press flash, call other party, wait for answer, press transfer, hangup. I believe that is what I saw on an earlier post. Doug ___ 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] Asterisk/linux 2.6.9 kernel build failure
Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. ___ 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] Is Busydetect obsolete in the latest CVS?
Perhaps I could pay you to teach me how to use google properly just send me Steve's terse post does actually contain the answer to your how to google question: http://www.google.com/search?q=%22ignoring+signalling%22+site%3Alists.digium.com Reverse engineering the above gives this: (entered in the search field on google) site:lists.digium.com ignoring signaling ___ 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] Asterisk not startin anymore.
Hello. I have this problem. In my asterisk box, I was running debian woody with asterisk package from backports.org. Last friday I upgraded from debian to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686, rebuild zaptel kernel module and also upgrade to asterisk 1.0.2. But now asterisk won't start. Here is more info #asterisk - (last lines) [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Nov 28 14:00:30 WARNING[1077059712]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device Nov 28 14:00:30 ERROR[1077059712]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Nov 28 14:00:30 ERROR[1077059712]: chan_zap.c:9130 setup_zap: Unable to register channel '1' Nov 28 14:00:30 WARNING[1077059712]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Nov 28 14:00:30 WARNING[1077059712]: loader.c:429 load_modules: Loading module chan_zap.so failed! When I modprobe wcfxs I got this: Nov 28 14:01:47 voiplab kernel: Freshmaker version: 71 Nov 28 14:01:47 voiplab kernel: Freshmaker passed register test Nov 28 14:01:50 voiplab kernel: Timeout waiting for calibration of module 0 Nov 28 14:01:52 voiplab kernel: Timeout waiting for calibration of module 0 Nov 28 14:01:52 voiplab kernel: Proslic Failed on Second Attempt to Auto Calibrate Nov 28 14:01:53 voiplab kernel: Proslic Failed on Second Attempt to Calibrate Manually. (Try -DNO_CALIBRATION in Makefile) Nov 28 14:01:53 voiplab kernel: Module 0: FAILED FXS (FCC) Nov 28 14:01:54 voiplab kernel: Module 1: Installed -- AUTO FXS/DPO Nov 28 14:01:54 voiplab kernel: Module 2: Installed -- AUTO FXO (FCC mode) Nov 28 14:01:55 voiplab kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 28 14:01:55 voiplab kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) It seems that the first FXS module of my TDM22B is broken. Is that correct? In that case how can I disable it? Just open the case and pull it out? Or can I apply a configuration parameter to disable it? Does this modules have a warranty? For how long? Thanx in advance. Andrés ___ 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] Registering on GK
Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos. ___ 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] Asterisk/linux 2.6.9 kernel build failure
Symlink /lib/modules/2.6.9/build to /usr/src/linux -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Sunday, November 28, 2004 10:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. ___ 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] OS Choice ?
Title: OS Choice ? Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.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] Phone Selection
Title: Phone Selection I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.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
RE: [Asterisk-Users] SetVar ALERT_INFO
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Its very helpful if you actually set them to something. The existence of
the variable isn't enough.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Sunday, November 28, 2004 4:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SetVar ALERT_INFO
Hello,
I've got my dialplan configured to do a double ring when a customer
service call comes in, and a normal ring when an extension is dialed
directly. When a customer service call is transferred, I want to ring
to revert back to normal.
In the local extension macro, I have the following
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=)
exten = s,n,SetVar(_ALERT_INFO=)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
When a call is transferred, here's what I see on the console
-- Executing NoOp(Zap/1-1, Bellcore-r3) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, ALERT_INFO=) in new stack
-- Executing SetVar(Zap/1-1, _ALERT_INFO=) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
It appears as though both ALERT_INFO and _ALERT_INFO are not set. This
isn't the case, however, because the party receiving the transferred
call still hears the double ring.
Now, I'm not sure if this is design or bug. I think I saw mention of
the _ doing something special but I'm not sure where and see no mention
of it in the wiki under SetVar.
Pointers would be appreciated.
Thanks,
Trevor Peirce
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RE: [Asterisk-Users] OS Choice ?
You always have a choice.. Gentoo, Debian... and as always RedHat is NOT an OS. It's a Distro. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Brecher Sent: Sunday, November 28, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OS Choice ? Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com http://www.successfulhosting.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
Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure
Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Sunday, November 28, 2004 10:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. shouldn't that be 'to /usr/src/linux-2.6' ___ 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] Registering on Gatekeeper
Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos ___ 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] Meetme Help !!!!
Oh stop messin with that conf file exten = 555,1,MeetMe(|dM) NEXT!!! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, November 27, 2004 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meetme Help Dave Cotton wrote: I would have: exten = 8600,1,Meetme(1234,M) just to have music on hold if there's only one person in the conference. Very cool! I'll give it a try, thanks! Doug ___ 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] Asterisk/linux 2.6.9 kernel build failure
On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote: Has anyone successfully built Asterisk with linux 2.6.9 kernel? Yes. It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. Someone posted a patch for the zaptel Makefile and it works fine. I've included a copy, I sorry I don't know who created it. Makefile.patch --- zaptel/Makefile.orig2004-10-14 10:24:35.497280408 -0400 +++ zaptel/Makefile 2004-10-14 11:02:09.561772322 -0400 @@ -65,6 +65,7 @@ PRIMARY=torisa #PRIMARY=wcfxo PWD=$(shell pwd) +KVER := $(shell uname -r) all: $(BUILDVER) @@ -72,8 +73,8 @@ linux26: linux26: prereq $(BINS) - @if ! [ -d /usr/src/linux-2.6 ]; then echo Link /usr/src/linux-2.6 to your kernel sources first!; exit 1 ; fi - make -C /usr/src/linux-2.6 SUBDIRS=$(PWD) modules + @if ! [ -d /lib/modules/$(KVER)/build ]; then echo Make sure that you have your kernel build environment at /lib/modules/$(KVER)/build; exit 1 ; fi + make -C /lib/modules/$(KVER)/build SUBDIRS=$(PWD) modules obj-m := $(MODULESO) ztdummy.o -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SetVar ALERT_INFO
On Sun, 28 Nov 2004, Brian West wrote:
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Its very helpful if you actually set them to something. The existence of
the variable isn't enough.
What the original poster said was that even though he set the variables to
nothing the observed effect was as if they still were set. And what you
are saying is that it is not enough to assign an empty value to the
variable to undo the effect of having set the ALERT_INFO?
Fair enough. If my unserstanding is correct perhaps someone can add a note
to the wiki? It is not totally obvious.
Peter
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Sunday, November 28, 2004 4:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SetVar ALERT_INFO
Hello,
I've got my dialplan configured to do a double ring when a customer
service call comes in, and a normal ring when an extension is dialed
directly. When a customer service call is transferred, I want to ring
to revert back to normal.
In the local extension macro, I have the following
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=)
exten = s,n,SetVar(_ALERT_INFO=)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
When a call is transferred, here's what I see on the console
-- Executing NoOp(Zap/1-1, Bellcore-r3) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, ALERT_INFO=) in new stack
-- Executing SetVar(Zap/1-1, _ALERT_INFO=) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
-- Executing NoOp(Zap/1-1, ) in new stack
It appears as though both ALERT_INFO and _ALERT_INFO are not set. This
isn't the case, however, because the party receiving the transferred
call still hears the double ring.
Now, I'm not sure if this is design or bug. I think I saw mention of
the _ doing something special but I'm not sure where and see no mention
of it in the wiki under SetVar.
Pointers would be appreciated.
Thanks,
Trevor Peirce
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Peter
--
Peter Svensson ! Pgp key available by finger, fingerprint:
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Remember, Luke, your source will be with you... always...
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[Asterisk-Users] Registering on Gatekeeper
Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos. ___ 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] Asterisk/linux 2.6.9 kernel build failure
Richard Lyman wrote: Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan Sent: Sunday, November 28, 2004 10:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. shouldn't that be 'to /usr/src/linux-2.6' ___ 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 Well, given the brain damaged nature of some distributions I have linux-2.4, linux-2.6 and linux all sym linked to linux-2.6.9 :-) And thanks to the suggestion originally given by Brian West my cvs of this AM compiles correctly. Thanks! ___ 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] Asterisk/linux 2.6.9 kernel build failure
Richard Lyman wrote: Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux shouldn't that be 'to /usr/src/linux-2.6' Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp (error with a reference to non-existent sk_buf-ethernet.mac or similar) but there is a specific patch for it in CVS :-) ___ 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] Asterisk/linux 2.6.9 kernel build failure
Andy Burns wrote: Richard Lyman wrote: Brian West wrote: Symlink /lib/modules/2.6.9/build to /usr/src/linux shouldn't that be 'to /usr/src/linux-2.6' Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp (error with a reference to non-existent sk_buf-ethernet.mac or similar) but there is a specific patch for it in CVS :-) ___ 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 Well, I am using my own distro - based on Linux From Scratch - so most of the distro-centric problems are not something I run into. The target platform for this will be a Soekris Net 4801 when I get it past the development phase (paperback book size computer). ___ 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] SetVar ALERT_INFO
On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote: Fair enough. If my unserstanding is correct perhaps someone can add a note to the wiki? It is not totally obvious. Peter, why don't *you* add a note to the Wiki? This is a community-supported project, and you're the community. ___ 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] [Fwd: Call Transfer between phones]
On Sun, 28 Nov 2004 11:28:05 -0500, Doug Lytle [EMAIL PROTECTED] wrote: Jeremy SALMON wrote: I have an PSTN line (X100P) and 10 grandstream budge tone phone. Jeremy, Receive call, press flash, call other party, wait for answer, press transfer, hangup. I believe that is what I saw on an earlier post. Doug Yes, assuming fairly recent firmware (1.0.5.16+). That does an attended transfer. To blind transfer you can just press transfer exten send and hangup. Mike ___ 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] SetVar ALERT_INFO
Brian West wrote:
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Its very helpful if you actually set them to something. The existence of
the variable isn't enough.
That's the problem - I only want it to be set for the FIRST call. The
SECOND call should forget about ALERT_INFO altogether and be processed
as though it was never set in the first place.
Is this not possible?
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RE: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure
I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Sunday, November 28, 2004 11:41 AM To: Asterisk List Subject: Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote: Has anyone successfully built Asterisk with linux 2.6.9 kernel? Yes. It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. Someone posted a patch for the zaptel Makefile and it works fine. I've included a copy, I sorry I don't know who created it. Makefile.patch --- zaptel/Makefile.orig2004-10-14 10:24:35.497280408 -0400 +++ zaptel/Makefile 2004-10-14 11:02:09.561772322 -0400 @@ -65,6 +65,7 @@ PRIMARY=torisa #PRIMARY=wcfxo PWD=$(shell pwd) +KVER := $(shell uname -r) all: $(BUILDVER) @@ -72,8 +73,8 @@ linux26: linux26: prereq $(BINS) - @if ! [ -d /usr/src/linux-2.6 ]; then echo Link /usr/src/linux-2.6 to your kernel sources first!; exit 1 ; fi - make -C /usr/src/linux-2.6 SUBDIRS=$(PWD) modules + @if ! [ -d /lib/modules/$(KVER)/build ]; then echo Make sure that you have your kernel build environment at /lib/modules/$(KVER)/build; exit 1 ; fi + make -C /lib/modules/$(KVER)/build SUBDIRS=$(PWD) modules obj-m := $(MODULESO) ztdummy.o -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure
Well, being a dinosaur (i.e. a very long UNIX/Linux experience person) I was not happy when it went from just a symlink of linux-kernel - linux to the current practice (RedHat style) of linux-kernel -linux-X.Y Just my .02$ Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Sunday, November 28, 2004 11:41 AM To: Asterisk List Subject: Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote: Has anyone successfully built Asterisk with linux 2.6.9 kernel? Yes. It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks. Someone posted a patch for the zaptel Makefile and it works fine. I've included a copy, I sorry I don't know who created it. Makefile.patch --- zaptel/Makefile.orig2004-10-14 10:24:35.497280408 -0400 +++ zaptel/Makefile 2004-10-14 11:02:09.561772322 -0400 @@ -65,6 +65,7 @@ PRIMARY=torisa #PRIMARY=wcfxo PWD=$(shell pwd) +KVER := $(shell uname -r) all: $(BUILDVER) @@ -72,8 +73,8 @@ linux26: linux26: prereq $(BINS) - @if ! [ -d /usr/src/linux-2.6 ]; then echo Link /usr/src/linux-2.6 to your kernel sources first!; exit 1 ; fi - make -C /usr/src/linux-2.6 SUBDIRS=$(PWD) modules + @if ! [ -d /lib/modules/$(KVER)/build ]; then echo Make sure that you have your kernel build environment at /lib/modules/$(KVER)/build; exit 1 ; fi + make -C /lib/modules/$(KVER)/build SUBDIRS=$(PWD) modules obj-m := $(MODULESO) ztdummy.o -- Dave Cotton [EMAIL PROTECTED] ___ 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] am i baned or something?
- Original Message - From: Joe Greco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 9:54 AM Subject: Re: [Asterisk-Users] am i baned or something? Sadly, the small portion of the Internet community that has a clue does not seem to care enough to do something to deal with this problem, such as finding ways to deliberately cause these mechanisms to break horribly until they're removed or turned off by their clueless MCSE admins. Regards, ... JG I think that would land you a nice quick trip to jail. ___ 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] isdn4linux delay
I have a weird problem and I cannot put my finger on it. I hope somebody can help me out. The quick way to solve this problem: Get an HFC-PCI card. It'll cost you 20-30 euros and with that you can use bristuff from http://junghanns.net/. This makes the HFC-PCI card a zaptel device. The other way is to dig into the linux kernel and try to hunt down the source of the problem, althoug I would rather use 20 euros on something else. isdn4linux sucks roy ___ 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] IP to IP call without server?
Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. ___ 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] IAX2 and FWD problems?
Hi,
I'm slowly getting to grips with *. My next quest is to get IAX2/FWD
calls working.
I've setup a fwd account and added IAX capability to it via the website.
I got the email saying it had been done with some welcome text and sample
phone numbers to try, such as 10001 for the answer phone.
I followed the setup example on the fwd site for configuring * to work
with fwd's IAX.
Basically when I make a call it rings out but no answer. I've tried
various numbers, 612, 613, 5 - all the same, no reply. So I guess
I have a problem at my end.
When I attempt a call I see:-
-- Executing SetCallerID(SIP/5061-084eda58, username) in new stack
-- Executing Dial(SIP/5061-084eda58,
IAX2/596146:[EMAIL PROTECTED]/612|60|r) in new stack
-- Called 526146:[EMAIL PROTECTED]/612
When the call is trying to connect:-
splat*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)65.39.205.121(None) 2/0
1/0 0ms ms ms UNKN
IAX2/65.39.205.121:4 65.39.205.121596146 3/0
2/0 0ms ms ms UNKN
2 active IAX channel(s)
And:
splat*CLI iax2 show registry
Host UsernamePerceived Refresh State
65.39.205.121:4569596146 Unregistered 60 Timeout
My section from extensions.conf is:
; Outgoing to FWD/IAX network, prefix calls with 7
exten = _777.,1,SetCallerId,${FWDCIDNAME}
exten = _777.,2,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _777.,3,Congestion
I'm using Asterisk 1.0.2
Any help would be very welcome!
Thanks
Mike
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[Asterisk-Users] Re: IAX2 and FWD problems?
Mike Dent [EMAIL PROTECTED] writes: I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. [...] Basically when I make a call it rings out but no answer. FWD's IAX gateway isn't working these days. Noone seems to know why. -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] IP to IP call without server?
check out skype - Original Message - From: nkb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 2:07 PM Subject: [Asterisk-Users] IP to IP call without server? Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. ___ 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] Asterisk/linux 2.6.9 kernel build failure
On Sun, 28 Nov 2004, Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? Havn't 2.6 adopted the /lib/modules/`uname -r`/build/ convention or something similar? Not having any 2.6-based machines online at the moment I can not check. This is from memory compiling out-of-tree modules a while back. Peter ___ 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] SetVar ALERT_INFO
On Sun, 28 Nov 2004, Chad Scott wrote: On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote: Fair enough. If my unserstanding is correct perhaps someone can add a note to the wiki? It is not totally obvious. Peter, why don't *you* add a note to the Wiki? This is a community-supported project, and you're the community. Because I do not know if I am correct or not. Because I do not have the equipment to verify if it is correct or not. It was a reminder to those involved to document their findings. Peter ___ 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] Asterisk/linux 2.6.9 kernel build failure
Peter Svensson wrote: On Sun, 28 Nov 2004, Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? Havn't 2.6 adopted the /lib/modules/`uname -r`/build/ convention or something similar? Not having any 2.6-based machines online at the moment I can not check. This is from memory compiling out-of-tree modules a while back. Peter ___ 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 Well - if 2.6.etc did adopt this it isn't reflected in actual make/make install world - i.e. nothing gets installed in /lib/modules/anywhere... And this is with kernel source from kernel.org - not a distro-tweaked source tree ___ 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] PRI Dialing failure?
So I reached the point where my PRI is accepting incoming calls, but I
cannot dialout. I must be doing something stupid, but I can't figure it
out. The Asterisk box is sitting between the Mitel and the phone company,
and has PRI lines to each. Asterisk was built from CVS r1-0
Log for a call from mitel heading outbound:
-
-- Accepting call from '' to '15123455476' on channel 0/14, span 2
Nov 28 15:27:05 DEBUG[1779]: chan_zap.c:1221 zt_enable_ec: No
echocancellation requested
-- Executing Dial(Zap/38-1, Zap/g3/5123455476) in new stack
Nov 28 15:27:05 NOTICE[1787]: app_dial.c:743 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy/congested at this time
Nov 28 15:27:05 DEBUG[1787]: app_dial.c:1029 dial_exec: Exiting with
DIALSTATUS=CHANUNAVAIL.
-- Executing Congestion(Zap/38-1, ) in new stack
The Configs:
-
zaptel.conf:
-
#Span 1 T1 to Remote Site
span=1,2,0,esf,b8zs
clear=1-24
#Span 2 Mitel
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
#Span 3 Verizon
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
-
zapata.conf:
-
[channels]
#Mitel
switchtype=dms100
context=frommitel
signalling=pri_net
group=1
channel = 25-47
#Verizon
switchtype=dms100
context=external
signalling=pri_cpe
group=2
channel = 49-71
-
extensions.conf
-
[globals]
OUTGOING-TRUNK=Zap/g3
[frommitel]
exten=1234,1,Playback(vm-goodbye)
exten=1234,2,Hangup
exten = _1NXXNXX, 1,Dial(${OUTGOING-TRUNK}/${EXTEN:1})
exten = _1NXXNXX, 2,Congestion
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Re: [Asterisk-Users] PRI Dialing failure?
On Sunday 28 November 2004 19:01, [EMAIL PROTECTED] wrote: So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 [...] - zapata.conf: - [channels] #Mitel switchtype=dms100 context=frommitel signalling=pri_net group=1 channel = 25-47 #Verizon switchtype=dms100 context=external signalling=pri_cpe group=2 channel = 49-71 - extensions.conf - [globals] OUTGOING-TRUNK=Zap/g3 Neither of your 2 PRIs are in group 3. Your span 1 seems non-existent. B ___ 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] Asterisk/linux 2.6.9 kernel build failure
On Sunday 28 November 2004 19:25, Steven P. Donegan wrote: Peter Svensson wrote: On Sun, 28 Nov 2004, Brian West wrote: I don't agree with this patch yet... It's the distro's fault for doing this wrong and I don't feel we have to work around it. The few people I talked to have Symlinks the build to /usr/src/linux or the like. Then again I may be wrong anyone know what is the right(tm) thing to do here is? Havn't 2.6 adopted the /lib/modules/`uname -r`/build/ convention or something similar? Not having any 2.6-based machines online at the moment I can not check. This is from memory compiling out-of-tree modules a while back. Well - if 2.6.etc did adopt this it isn't reflected in actual make/make install world - i.e. nothing gets installed in /lib/modules/anywhere... And this is with kernel source from kernel.org - not a distro-tweaked source tree 2.6 did adopt it. Look at the target _modinst_: in the Linux top level Makefile. B ___ 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] IP to IP call without server?
[EMAIL PROTECTED] wrote: Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. The answer is both Yes and No: ~~ Yes: ~~ If the two IP endpoints can connect to each other, and the protocols are compatible, peer-to-peer VoIP connections are totally feasable. Technically speaking, most VoIP protocols are peer-to-peer; as long as both ends are using the same protocol, any call can ultimately be a direct connection between endpoints. If you can configure the dialplan in your terminal (IP phone or soft client) to know exactly the IP address of the destination endpoint (your buddy's IP phone or soft client), and the destination endpoint is willing to accept incoming connections from your client, you're good to go. Getting this going across networks is usually a bit more complicated that that. ~~ No: ~~ Being really new, you'd avoid a ton of frustration for yourself (not to mention a steep learning curve) if you interface through some sort of registration server. Why create needless hassle for yourself? When you start adding firewalls, NATs, DHCP, mobility and a bunch of other things, it becomes attractive to centralize these configurations in a machine that provides lookup services to all users. It's not *needed*, but it is very beneficial. Ultimately, the complexity of the network is such that it will become a necessity. For example, the internet works fine without DNS (some apps do not, but I digress), but without it, everyone would need to keep track of IP addresses. DNS has become essential. Similarly, there is no technical reason to have Google, but can you imagine surfing without it? If you want to use VoIP without getting into Asterisk, services such as FreeWorldDialup are worth looking into. Cheers, Jim. ___ 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] IP to IP call without server?
nkb wrote: Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? I think this kind of functionality is supported by some SIP phones. Well the docs of my budgetone says it does, but I've never tried it. Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? From what I understand, if you are using, say, 2 IAXys, then they will sort of automagically communicate directly with each other once the communication is established. I also think SIP has a similar mechanism (REINVITE?) - but then again I'm fairly new to VoIP so I might be talking all rubbish. If I am please someone correct me :-) Cheers, Jean-Michel. ___ 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] PRI Dialing failure?
*Smack*, you're right, changing the g3 to g2 help nicely. But now the PRI seems to be refusing the call (Channel 0/1 got hangup): --snip-- -- Executing Answer(Zap/38-1, ) in new stack -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 16:08:14 DEBUG[1894]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Called g2/15123455476 -- Channel 0/1, span 3 got hangup Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:2465 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/49-1 Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:1975 zt_hangup: Hangup: channel: 49 index = 0, normal = 43, callwait = -1, thirdcall = -1 Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:2103 zt_hangup: Already hungup... Calling hangup once, and clearing call Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:2377 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/49-1 Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:1193 update_conf: Updated conferencing on 49, with 0 conference users Nov 28 16:08:15 DEBUG[1900]: chan_zap.c:2459 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/49-1 -- Hungup 'Zap/49-1' == No one is available to answer at this time On Sunday 28 November 2004 19:01, [EMAIL PROTECTED] wrote: So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 [...] - zapata.conf: - [channels] #Mitel switchtype=dms100 context=frommitel signalling=pri_net group=1 channel = 25-47 #Verizon switchtype=dms100 context=external signalling=pri_cpe group=2 channel = 49-71 - extensions.conf - [globals] OUTGOING-TRUNK=Zap/g3 Neither of your 2 PRIs are in group 3. Your span 1 seems non-existent. B ___ 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] Asterisk/linux 2.6.9 kernel build failure
On Sun, 28 Nov 2004, Bob Goddard wrote: On Sunday 28 November 2004 19:25, Steven P. Donegan wrote: Well - if 2.6.etc did adopt this it isn't reflected in actual make/make install world - i.e. nothing gets installed in /lib/modules/anywhere... And this is with kernel source from kernel.org - not a distro-tweaked source tree 2.6 did adopt it. Look at the target _modinst_: in the Linux top level Makefile. A look at the The Linux Kernel Module Programming Guide seems to confirm it as at least a recommended way of doing things. http://www.tldp.org/LDP/lkmpg/2.6/html/x419.html In our Asterisk build script we first check for /lib/modules/ and if the includes are not present there we fall back to /usr/src/linux*/. Peter ___ 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] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? Any reply appreciated. Tomek ___ 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] asterisk based bbs
In data Sun, 28 Nov 2004 16:55:06 +0100, Michael Vogel [EMAIL PROTECTED] ha scritto: lenz schrieb: I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about asterisk, but a vocal bbs that runs on asterisk, so that people can call, hear the discussion of the day, leave messages, etc. It doesn't really make sense to me. It only makes sense for some very limited fields. e.g. when somebody cannot read or write (he hasn't learned it or is blind). Everybody else could use the computer. Most people who would use such a system do have a computer at home I guess. yes, but it could be a funny thing to do. I have been a long time bbs enthusiast and would have loved - back in the beginning of the 90s, with the first 14400 voice modems - to run a voice only bbs, with people talking instead of typing. now with * it is not only feasible, but by linking up to some free phone number provider could be free for everyone to use. so I thought this idea would come up in somebody else's mind :-) l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] PRI Dialing failure?
[EMAIL PROTECTED] wrote: *Smack*, you're right, changing the g3 to g2 help nicely. But now the PRI seems to be refusing the call (Channel 0/1 got hangup): --snip-- -- Executing Answer(Zap/38-1, ) in new stack -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 16:08:14 DEBUG[1894]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Called g2/15123455476 -- Channel 0/1, span 3 got hangup Try turning PRI debug on and looking at that... post back if you still can't figure it out. Nick ___ 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 EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Tomasz Chmielewski wrote: Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. ___ 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 EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. Or off course you can buy digium cards. They look pretty cool anyway - can't wait to receive the onces I have ordered :-) ___ 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] OS Choice ?
Which Distro is the most commonly used distro with Asterisk please ? Best Regards, Alex Brecher -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Sunday, November 28, 2004 12:37 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OS Choice ? You always have a choice.. Gentoo, Debian... and as always RedHat is NOT an OS. It's a Distro. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Brecher Sent: Sunday, November 28, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OS Choice ? Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com http://www.successfulhosting.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
RE: [Asterisk-Users] Phone Selection
Title: Phone Selection Anybody here have suggestions on these phones please ? Best Regards, Alex Brecher From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex BrecherSent: Sunday, November 28, 2004 12:36 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Phone Selection I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.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
Re: [Asterisk-Users] OS Choice ?
See the reply below yours. I would hazard a guess that Redhat and SuSE, followed by Debian, are probably the top three (RH and SuSE because of market share, and enterprise server distros thbey have). Greg Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? Best Regards, Alex Brecher -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Sunday, November 28, 2004 12:37 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OS Choice ? You always have a choice.. Gentoo, Debian... and as always RedHat is NOT an OS. It's a Distro. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Brecher Sent: Sunday, November 28, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OS Choice ? Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com http://www.successfulhosting.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
Re: [Asterisk-Users] PRI Dialing failure?
This looks like a config issue, class of service barred but getting config information out of verizon is nearly impossible. I compared what the Mitel is sending to asterisk (since the mitel does work with the PRI) with what asterisk is sending and do not see any large differences. debugging spam 3 asterisk to verizon: --snip-- Enabled debugging on span 3 -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Making new call for cr 32770 Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Number not available (67) '' ] [70 0c a1 31 35 31 32 33 34 35 35 34 37 36] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '15123455476' ] -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 17:04:29 DEBUG[2287]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Called g2/15123455476 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32770/0x8002) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 b6] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Incoming call barred (54), class = Service or Option not Available (3) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 3 got hangup Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2465 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/49-1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:1975 zt_hangup: Hangup: channel: 49 index = 0, normal = 43, callwait = -1, thirdcall = -1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2103 zt_hangup: Already hungup... Calling hangup once, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2377 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/49-1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:1193 update_conf: Updated conferencing on 49, with 0 conference users Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2459 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/49-1 -- Hungup 'Zap/49-1' == No one is available to answer at this time Nov 28 17:04:29 DEBUG[2293]: app_dial.c:1029 dial_exec: Exiting with DIALSTATUS=NOANSWER. -- Executing Congestion(Zap/38-1, ) in new stack - debugging span 2, mitel to asterisk --- Enabled debugging on span 2 Protocol Discriminator: Q.931 (8) len=29 Call Ref: len= 2 (reference 7795/0x1E73) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 8e] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 14 ] [70 0c a1 31 35 31 32 33 34 35 35 34 37 36] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '15123455476' ] -- Making new call for cr 7795 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 40563/0x9E73) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8e] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 14 ] -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 17:07:42 DEBUG[2309]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Called g2/15123455476 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 40563/0x9E73) (Terminator) Message
Re: [Asterisk-Users] OS Choice ?
Do I have any other options besides RH 9.0 ? You always have a choice. Most distros provide some form of download for their media. RH/FC, regardless of version, is easiest IMO because of simple ISO image availability. If you really wanted, you could build up a Linux machine based only on a kernel, bootstrap a GCC build, and build everything else you need from there. I've done it before, and that's why I prefer to download ISO images, burn CDs, and install the distro. Greg ___ 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 EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Jean-Michel Hiver wrote: Tomasz Chmielewski wrote: Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. too bad. I have dozens of these EICON Diva cards, I thought I could use them and not buy any additional hardware :( Tomek ___ 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] asterisk based bbs
[EMAIL PROTECTED] wrote: In data Sun, 28 Nov 2004 16:55:06 +0100, Michael Vogel [EMAIL PROTECTED] ha scritto: lenz schrieb: I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about asterisk, but a vocal bbs that runs on asterisk, so that people can call, hear the discussion of the day, leave messages, etc. It doesn't really make sense to me. It only makes sense for some very limited fields. e.g. when somebody cannot read or write (he hasn't learned it or is blind). Everybody else could use the computer. Most people who would use such a system do have a computer at home I guess. yes, but it could be a funny thing to do. I have been a long time bbs enthusiast and would have loved - back in the beginning of the 90s, with the first 14400 voice modems - to run a voice only bbs, with people talking instead of typing. now with * it is not only feasible, but by linking up to some free phone number provider could be free for everyone to use. so I thought this idea would come up in somebody else's mind :-) l. It's a very interesting idea. The more I think about it, the more I wonder . . . I like it somehow, but I can't quite picture how people would use it. Probably there'd need to be some sort of GUI so that threads could be identified and sorted. Each message would contain an audio recording instead of text, so you'd pick the thread, connect via your media device and listen/post using your audio connection coupled with the GUI. A few questions come to mind: How would you quote? Top post vs. bottom post? I think a new paradigm would be needed, but the technology and the network are certainly up to it. And just think of how much more exciting flame wars could get! One could affect a Winston Churchill accent and really let loose! And forget smileys -- Now there's sound effects!! Very visionary thinking. Whether it's a good idea or not? Only time will tell. ___ 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] Voicemail configuration via PostgreSQL?
I've been looking at the wiki and the source for a long time now and I just can't seem to get this straight... is VM config by PostgreSQL functional? From what I've seen it looks like it isn't. I've noticed: 1) Setting USE_POSTGRES_VM_INTERFACE to 1 in apps/Makefile sets CFLAGS to include -DUSEPOSTGRESVM and adds -lpq to the compiler's command line. 2) USEPOSTGRESVM isn't mentioned anywhere in the source except for the above Makefile. 3) libpq isn't mentioned anywhere except cdr_pgsql.c (and CDRs work with PostgreSQL on my machine) and app_sql_postgres.c. Are these signs all red herrings? Does this actually work for anyone? I'm using CVS HEAD as of a few hours ago, BTW. Thanks, Tim Mattison [EMAIL PROTECTED] ___ 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] Registering on Gatekeeper
If you are using GnuGK, i think this should do, in your h323.conf file, configure an asterisk endpoint as follow for instance [time] Username type=h323 e164=99 context=test K. - Original Message - From: Nahuel Alejandro Ramos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 5:52 PM Subject: [Asterisk-Users] Registering on Gatekeeper Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos. ___ 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] How to test if PCI 2.2?
On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro [EMAIL PROTECTED] wrote: Only way that I know is to open the case and look at the slot to see if there are two dividers. I would be interested in knowing this as well. I've seen many motherboards that claim to be PCI 2.2 compliant, but they have only one divider in the PCI slots. (example: Asus A7N8X) Or were you saying that it was PCI 2.3 slots that have two dividers? A related question... Do TDM400P cards work OK in PCI 2.1 slots? The datasheet says available PCI slot and PCI 2.2 compliant but I see no hard requirement... just want to be sure before I deploy an idle PCI 2.1 board I have here. -- Lee ___ 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] PRI Dialing failure?
This looks like a config issue, class of service barred but getting config information out of verizon is nearly impossible. I compared what the Mitel is sending to asterisk (since the mitel does work with the PRI) with what asterisk is sending and do not see any large differences. debugging spam 3 asterisk to verizon: --snip-- Enabled debugging on span 3 -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Making new call for cr 32770 Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Number not available (67) '' ] [70 0c a1 31 35 31 32 33 34 35 35 34 37 36] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '15123455476' ] -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 17:04:29 DEBUG[2287]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Called g2/15123455476 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32770/0x8002) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 b6] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Incoming call barred (54), class = Service or Option not Available (3) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 3 got hangup Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2465 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/49-1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:1975 zt_hangup: Hangup: channel: 49 index = 0, normal = 43, callwait = -1, thirdcall = -1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2103 zt_hangup: Already hungup... Calling hangup once, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2377 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/49-1 Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:1193 update_conf: Updated conferencing on 49, with 0 conference users Nov 28 17:04:29 DEBUG[2293]: chan_zap.c:2459 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/49-1 -- Hungup 'Zap/49-1' == No one is available to answer at this time Nov 28 17:04:29 DEBUG[2293]: app_dial.c:1029 dial_exec: Exiting with DIALSTATUS=NOANSWER. -- Executing Congestion(Zap/38-1, ) in new stack - debugging span 2, mitel to asterisk --- Enabled debugging on span 2 Protocol Discriminator: Q.931 (8) len=29 Call Ref: len= 2 (reference 7795/0x1E73) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 8e] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 14 ] [70 0c a1 31 35 31 32 33 34 35 35 34 37 36] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '15123455476' ] -- Making new call for cr 7795 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 40563/0x9E73) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8e] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 14 ] -- Executing Dial(Zap/38-1, Zap/g2/15123455476) in new stack -- Accepting call from '' to '15123455476' on channel 0/14, span 2 Nov 28 17:07:42 DEBUG[2309]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested -- Called g2/15123455476 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 40563/0x9E73) (Terminator) Message
Re: [Asterisk-Users] PRI Dialing failure?
On Sun, 28 Nov 2004 [EMAIL PROTECTED] wrote: This looks like a config issue, class of service barred but getting config information out of verizon is nearly impossible. I compared what the Mitel is sending to asterisk (since the mitel does work with the PRI) with what asterisk is sending and do not see any large differences. Perhaps they dislike the numbering plan for the calling number you sent? The Mitel sends no calling number in the debug log while Asterisk sent an empty number with TON/NPI unknown/unknown. Just an idea. Peter ___ 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 test if PCI 2.2?
On Sun, 28 Nov 2004, Lee wrote: On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro [EMAIL PROTECTED] wrote: Only way that I know is to open the case and look at the slot to see if there are two dividers. I would be interested in knowing this as well. I've seen many motherboards that claim to be PCI 2.2 compliant, but they have only one divider in the PCI slots. (example: Asus A7N8X) Or were you saying that it was PCI 2.3 slots that have two dividers? There are no physical differences between pci 2.1 and pci 2.2. See http://www.pcisig.com/news_room/faqs/pci_sig_faq.pdf (2nd last clause) and http://www.pcisig.com/specifications/conventional/conventional_pci/2_2sum1215.pdf Apparently 2.2 is mostly a clarification/correction of 2.1. Peter ___ 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] IP to IP call without server?
I agree you can do this with SIP. but I would use skype, msn, yahoo or VOIP blasters (get on ebay) for a simple call to call without a server. its too much effort and too much to learn for a simple call. On Mon, 29 Nov 2004 04:07:43 +0900, nkb [EMAIL PROTECTED] wrote: Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. ___ 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 -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.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] soxmix
Does soxmix works with asterisk ver. 0.9?
I have ver. sox-12.17.5 on Gentoo but the option m does not combine
two WAV files (In and Out) into one file. I have two separate files
in /monitor folder.
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 711,2,Monitor(wav,${CALLFILENAME},m)
exten = 711,3,Dial(${sales_support},20,r)
exten = 711,4,Voicemail(u11); Right to voicemail
exten = 711,5,Hangup()
--
#Joseph
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Re: [Asterisk-Users] Voicemail configuration via PostgreSQL?
After lots and lots of digging it appears to me as if support for DB based configuration is only via ODBC (known as extconfig). Sorry for the noise. I'll be sure to compile my findings on my site and post a suitable followup when it is complete. On Sun, 2004-11-28 at 16:31 -0500, Tim Mattison wrote: I've been looking at the wiki and the source for a long time now and I just can't seem to get this straight... is VM config by PostgreSQL functional? From what I've seen it looks like it isn't. I've noticed: 1) Setting USE_POSTGRES_VM_INTERFACE to 1 in apps/Makefile sets CFLAGS to include -DUSEPOSTGRESVM and adds -lpq to the compiler's command line. 2) USEPOSTGRESVM isn't mentioned anywhere in the source except for the above Makefile. 3) libpq isn't mentioned anywhere except cdr_pgsql.c (and CDRs work with PostgreSQL on my machine) and app_sql_postgres.c. Are these signs all red herrings? Does this actually work for anyone? I'm using CVS HEAD as of a few hours ago, BTW. Thanks, Tim Mattison [EMAIL PROTECTED] ___ 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 Selection
I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? I briefly tested the 480i a couple of weeks ago. Had a problem in that it would not use the tftp server address contained in the dhcp response, so had to define everything from the keypad to make it work. The quality of the audio was good, the speakerphone function worked, and all other very basic phone functions that I tried (not an expensive test at all) worked as expected. There seemed to be a lot of this function will be implemented in a later software release kind of thing going on. I did not write down the s/w version that it was running, but I do remember there were two additional releases available after the one I had. I would not deploy this phone in large quantities at this time as they would be a support nightmare. For small quantities, not a bad phone at all. That's about all I can tell you on it. I use a 7960 for day to day business use and like it very well. It feels like a phone, works like a phone, excellent speakerphone, and continues to function well. Probably a little over priced these days. I'll stay with it for now. You should probably dig through the wiki as I'm sure there is more detail there on lots of different phones. Rich ___ 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] soxmix
You may want to try calling StopMonitor to see if that forces a merge.
I've used Monitor before on Gentoo and it works with soxmix but I've
never tried to do it without an explicit StopMonitor.
On Sun, 2004-11-28 at 15:36 -0700, Joseph wrote:
Does soxmix works with asterisk ver. 0.9?
I have ver. sox-12.17.5 on Gentoo but the option m does not combine
two WAV files (In and Out) into one file. I have two separate files
in /monitor folder.
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 711,2,Monitor(wav,${CALLFILENAME},m)
exten = 711,3,Dial(${sales_support},20,r)
exten = 711,4,Voicemail(u11); Right to voicemail
exten = 711,5,Hangup()
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[Asterisk-Users] OT: mixing monitor files to stereo wav
Hi, i am looking for a tool to merge the two wav files of a monitored call into one. soxmix does that well but actually merges the two channels. I would prefer a solution that creates a stereo wav file of the two mono files so you have the called party on one (e.g. left) channel and the calling party on the other (e.g. right). I can do this interactivly using audacity but i am looking for a tool to automate this. Anybody knows a piece of open software that supports this? Thanks in advance Stefan ___ 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 Selection
On Sun, 28 Nov 2004 16:32:10 -0600, Rich Adamson [EMAIL PROTECTED] wrote: I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? I briefly tested the 480i a couple of weeks ago. Had a problem in that it would not use the tftp server address contained in the dhcp response, so had to define everything from the keypad to make it work. The quality of the audio was good, the speakerphone function worked, and all other very basic phone functions that I tried (not an expensive test at all) worked as expected. There seemed to be a lot of this function will be implemented in a later software release kind of thing going on. I did not write down the s/w version that it was running, but I do remember there were two additional releases available after the one I had. I would not deploy this phone in large quantities at this time as they would be a support nightmare. For small quantities, not a bad phone at all. That's about all I can tell you on it. I use a 7960 for day to day business use and like it very well. It feels like a phone, works like a phone, excellent speakerphone, and continues to function well. Probably a little over priced these days. I'll stay with it for now. You should probably dig through the wiki as I'm sure there is more detail there on lots of different phones. Rich Someone once told me that he would never consider using a SIP phone unless it had been through several software releases / revisions. In my experience, this kind of thinking seems to work well. For example, this 480i is relatively new to the market, having only been out less than a year. Bugs are probably still being worked out, features still to be implemented, and with those features, more bugs. I'd recommend the Cisco 7940/7960 series phones. Another phone I've been impressed with so far, although haven't tested extensively is the Polycom Soundpoint IP500. It seems to be a solid phone, with a feature set that gives Cisco a run for its money. ___ 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] soxmix
Joseph wrote:
Does soxmix works with asterisk ver. 0.9?
I have ver. sox-12.17.5 on Gentoo but the option m does not combine
two WAV files (In and Out) into one file. I have two separate files
in /monitor folder.
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 711,2,Monitor(wav,${CALLFILENAME},m)
exten = 711,3,Dial(${sales_support},20,r)
exten = 711,4,Voicemail(u11); Right to voicemail
exten = 711,5,Hangup()
I belive you need 1.0 for the m option to work.
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Re: [Asterisk-Users] soxmix
On Sun, 2004-11-28 at 17:43 -0500, Tim Mattison wrote:
You may want to try calling StopMonitor to see if that forces a merge.
I've used Monitor before on Gentoo and it works with soxmix but I've
never tried to do it without an explicit StopMonitor.
I've tried make not difference:
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 711,2,Monitor(wav,${CALLFILENAME},m)
exten = 711,3,Dial(${sales_support},20,r)
exten = 711,4,StopMonitor
exten = 711,5,Voicemail(u11) ; Right to voicemail
exten = 711,6,Hangup()
Sill ending up with two files:
-rw-r--r-- 1 root root 269804 Nov 28 16:16 11-20041128-161619|m-in.wav
-rw-r--r-- 1 root root 254924 Nov 28 16:16 11-20041128-161619|m-out.wav
Though that pipe +m is part of the filename, as if the soxmix wasn't
executing. Soxmix is in the path.
--
#Joseph
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Re: [Asterisk-Users] How to test if PCI 2.2?
Lee wrote: On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro [EMAIL PROTECTED] wrote: Only way that I know is to open the case and look at the slot to see if there are two dividers. I would be interested in knowing this as well. I've seen many motherboards that claim to be PCI 2.2 compliant, but they have only one divider in the PCI slots. (example: Asus A7N8X) Or were you saying that it was PCI 2.3 slots that have two dividers? The motherboard has PCI slots with one divider with a smaller part away from the chasis. Do you want to say that the ASUS A7N8X says it is PCI 2.2 but it is actually not A related question... Do TDM400P cards work OK in PCI 2.1 slots? The datasheet says available PCI slot and PCI 2.2 compliant but I see no hard requirement... just want to be sure before I deploy an idle PCI 2.1 board I have here. I have the card installed on a MSI 865PE/G Neo2 board. I found the motherboard manual / description (MSI 865PE/G) on the web (http://www.msi.com.tw/program/products/mainboard/mbd/pro_mbd_detail.php?UID=454): It says Six 32-bit v2.3 Master PCI bus slots (support 3.3v/5v PCI bus interface). I got the hint to use cat /proc/pci |grep PCI bridge The result is just: PCI bridge: Intel Corp. 82865G/PE/P Processor to AGP Controller (rev 2). PCI bridge: Intel Corp. 82801BA/CA/DB/EB PCI Bridge (rev 194). PCI bridge: Digital Equipment Corporation DECchip 21152 (rev 3). I have compiled the modules and they are added into the /etc/modules.conf and added into /etc/modeprobe.conf. When I restart asterisk with zapata.conf: context=internal signalling=fxo_ls immediate=no busydetect=no echocancel=yes callerid=Internal Line 1 603 channel = 1 Asterisk dies with unable to specify channel 1: No such device or address When I try ztfg -vv I get: Zaptel Configuration === Channel map: 0 channels configured. lsmod shows: wctdm zaptel /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 2: Installed -- AUTO FXO (FCC mode) Nov 27 00:37:56 dns kernel: Module 3: Installed -- AUTO FXO (FCC mode) What can I do next? bye Ronald ___ 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] soxmix
On Sun, 2004-11-28 at 18:11 -0500, Dave DeChellis wrote: Joseph wrote: Does soxmix works with asterisk ver. 0.9? [snip] I belive you need 1.0 for the m option to work. That was my initial impression. So need to pull few unstable packages from portage to compile 1.0.1 or 1.0.2 -- #Joseph ___ 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] IP to IP call without server?
On Sun, 28 Nov 2004 17:25:25 -0500, Steven Kalcevich (Lists) wrote: I agree you can do this with SIP. but I would use skype, msn, yahoo or VOIP blasters (get on ebay) for a simple call to call without a server. its too much effort and too much to learn for a simple call. I'm not a big fan of supporting proprietary soltuions so I'd avoid Skype. However, what about Free World Dialup? Uses common sip clients, they have the new Pulver communicator which supports video, voice, and text chat. Seems like a good solution. Michael On Mon, 29 Nov 2004 04:07:43 +0900, nkb [EMAIL PROTECTED] wrote: Hi. I'm really new. I was just wondering if it is possible at all to do a IP to IP call without a * server (or as a matter of fact, any other kind of server)? say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we all both be registered with the same server to do that? Can this not be done without passing thru server (*)? Thanks. ___ 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 -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 test if PCI 2.2?
On November 27, 2004 08:53 pm, Steve Totaro wrote: Only way that I know is to open the case and look at the slot to see if there are two dividers. I would be interested in knowing this as well. What exactly do you mean by two dividers? Almost every PCI motherboard I have has only one, and they're *all* PCI 2.2. Smaller part of the slot at the back: 5V PCI slot, smaller part to the front, 3.3V PCI, but all PCI 2.2. -A. ___ 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] Entire mailing list archive download?
A while back, I found a site that had the entire asterisk-users mailing list archive in mbox format. Does anybody know if and where such a thing is availible? PJ -- All men know the utility of useful things; but they do not know the utility of futility. -- Chuang-tzu ___ 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] PRI Dialing failure?
I figured it out. The key was that the mitel was always dialing channel 14, Asterisk was always dialing channel 1. When I reconfigure asterisk to round robin dial the PRI group it started at 1 and worked upwards. If the channel in the debug output is 9 dialing works. It appears that the bottom 8 of the PRI channels are DID only. So by changing my zapata.conf to this: #Verizon switchtype=national context=external signalling=pri_cpe #Channels 49-56 are DID only group=2 channel = 49-56 group=3 channel = 57-71 and setting the extensions.conf to dial group 3 yt works... Thanks for the tip to the PRI debug output. Mark Farver On Sun, 28 Nov 2004 [EMAIL PROTECTED] wrote: This looks like a config issue, class of service barred but getting config information out of verizon is nearly impossible. I compared what the Mitel is sending to asterisk (since the mitel does work with the PRI) with what asterisk is sending and do not see any large differences. Perhaps they dislike the numbering plan for the calling number you sent? The Mitel sends no calling number in the debug log while Asterisk sent an empty number with TON/NPI unknown/unknown. Just an idea. Peter ___ 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] Asterisk not startin anymore.
Andres Junge wrote: snip It seems that the first FXS module of my TDM22B is broken. Is that correct? In that case how can I disable it? Just open the case and pull it out? Or can I apply a configuration parameter to disable it? You should be able to do so by removing all reference to that particular module in /etc/zaptel.conf and /etc/asterisk/zapata.conf, without having to pull the module out. although I remembered having a bum FXO module on my TDM22B, but it didn't cause the problem you encountered. might have been a different hardware problem though. Does this modules have a warranty? For how long? IIRC, all Digium hardware should have a one-year warranty on it, although you may have to check with the people you bought it from. flynn ___ 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] Registering on Gatekeeper
I have already tryed this but asterisk always send root as h323_id On Sun, 28 Nov 2004 21:31:52 -, kido noagbodji [EMAIL PROTECTED] wrote: If you are using GnuGK, i think this should do, in your h323.conf file, configure an asterisk endpoint as follow for instance [time] Username type=h323 e164=99 context=test K. - Original Message - From: Nahuel Alejandro Ramos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 5:52 PM Subject: [Asterisk-Users] Registering on Gatekeeper Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos. ___ 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
