Re: [Asterisk-Users] SIP/grandstream not registering
It looks like you have you * on public IP and your phones on private, most likely behind NAT if so in your sip.conf under each [grandstreamX] you most likely need: nat=yes - Original Message - From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 1:44 AM Subject: [Asterisk-Users] SIP/grandstream not registering hi, i can't seem to register my grandstream SIP to * server... i have my grandstream IP as 192.168.0.11 want to register to * at 202.51.xx.xxx. sip show peers says that my grand stream has unspecified IP but when i try to dial a number it gets this error... WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 40939 (Response) my sip.conf is... [grandstream2] type=peer host=dynamic secret=grandstream2 reinvite=no canreinvite=no qualify=60 [grandstream2] type=user host=dynamic secret=grandstream2 context=outgoing reinvite=no canreinvite=no qualify=60 help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Slow wiki?
www.voip-info.org had the most visitors ever today (Jan 2). Slowed down a little, but otherwise no problems. The Tiki Wiki software is rather resource greedy, its on a dedicated server, but it does LOTS of database queries for each page it displays. Jim James H. Thompson [EMAIL PROTECTED] Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 02, 2004 4:06 AM Subject: Re: [Asterisk-Users] Slow wiki? On Fri, 2004-01-02 at 07:59, Philipp von Klitzing wrote: Hi there, is this a problem with the Wiki software or the DB? The delay is still tolerable, but not exactly nice to work with. http://www.voip-info.org/tiki-index.php?page=Asterisk+billing Page generated in: 2.35 seconds Wiki software has a bad reputation of too many queries and there for always slow operations. Of course when I hit that URL this morning it listed as 1.1 seconds. It may well be that the recent mention is slashdot may have had increased the visitors to the site at that time. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gotoif / last called
Hi
Ive been trying to get this to work for ages now, basicaly im trying
to do if ${woteva} = (nothing), or its none existenant then do
label 1, else label 2. for my last called function, so it will play a
different message if theres no last call in the system or it was
anonymous. ive tried
There is a way to test for an empty string:
GotoIf($[${CALLERIDNUM} = ]?3:2)
exten = 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ]?4:5)
and heaps of other combinations, most times it ses syntax error when
im in asterisk verbose
My guess is that this won't work. Instead look into DBput and DBget and
do something like
DBput(last-call/${CALLERIDNUM}=value)
The advantage of this is, by the way, that the last caller data will also
be available after a restart of Asterisk.
Also say in my example above, its true so it goes to 4, does that mean
it goes to extension 4.. or rule 4 in that dialplan ?
You'll jump to priority 4, not extension 4.
oh while im asking.. is there a way to make global changes to a
context
Look at Macros.
Cheers, Philipp
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Re: [Asterisk-Users] Call recording/SIP not loggin IN
Hi! [xlite1] type=user Make this [xlite1user] Adjust your extension.conf accordingly. [xlite1] type=peer Make this [xlite1peer] Adjust your extension.conf accordingly. The alternative is to merge both entries and use type=friend instead. my grandstream is also not registering to *. You expect us to guess your SIP setup for the GS? :- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
SW: Thanks a million for the statement that I only need these two files and they can be just about empty ! David Carter: many thanks for those files which I will study Rich Adamson: That is so re-assuring! That may sound odd but its realy helpful to have the problems I am facing acknowledged and makes me feel that others really see the need for, in effect, intuitive docs to get the novice on-board. I used to write code, now I leave it to my staff, but I guess I can go there. What I am doing is evaluating * to see if we as a company should use and support it rather than just buying in Quintum boxes or whatever. No doubt many others are doing the same. As a company we write software for end-users and I insist that an average 16 year old must be able to make it work, at the basic level, without grief - it must be intuitive. OK make that an average linux administrator for */VOIP but again it really needs to be intuitive - but I guess I am preaching to the convereted. I would like to offer to try and do that in the wiki - but realistically I don't have the time. Still I am feeling a bit guilty now having got such solid support. Thank you. John - John A Coll, Director, Connection Software 391 City Road, LONDON, EC1V 1NE, UK Tel: 020 7713 8000 From outside UK Tel: +44 20 7713 8000 Fax: 020 7713 8001 Fax: +44 20 7713 8001 Email: [EMAIL PROTECTED] Web: www.csoft.co.uk PGP Public Key from keyserver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 03 January 2004 01:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :) This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: snip I've been at this off and on for two weeks Linux admin and firewalls seem trivial compared to this so I must be missing something pretty basic :) Careful, that's the wrong thing to say on this list; but, the exact same thing has been reiterated at least several thousand times (minimum) in the last few months. The underlying problem truly is that even for those of us that have been professionally involved with telephony (for years), the initial learning curve for * is far steeper then the average implementor can begin to comprehend. Please folks, let's not start the _weekly_ read the code/docs war once again; for the experienced ones that really want to click on reply, please don't The bottom line is that unless you can read/comprehend code rather quickly, the technical documentation does not exist in any reasonable form. Lots of very good people are trying very very hard, but the fact is that far more technical doc exists only in the code then one would expect from such an excellent application. (The subject really has been covered in very negative terms many times, if one can find it. One of the better choices for newbie research really is http://www.voip-info.org/tiki-index.php , but even this is very much a 'work-in-progress'. That's a Good Thing!!! There is also a fair number of folks on the list that are trying to earn a living via * that won't take the time to respond to even the most basic questions for obvious reasons. Their signatures will become very apparent.) Not all of the documentation problem is really related to *; there really is a lot of interpretation/advancement/research going on with SIP vendors that frequently initiate postings related to problems/comments on the list. Once you get a basic * system working, you'll find significant issues with the SIP standards in terms of NAT and many many other items. That's not putting * down, its just the nature of non-commercial internet standards. I do believe that most implementors find the /usr/src/asterisk/README.* to be helpful, and some other directories that contain sample configs (of which the directory names are so unobvious I can't find them after a couple of beers. ;) You will find that not all SIP vendors interpret the exact same standards in the same way. For those of us that have tried, software/hardware SIP phones vary dramatically in terms of interoperability with * (and other telephony apps). Some get it reasonably right, and other vendors try to advance the standards with their own interpretations. And, a few are obviously basement operations with minimal informed staff. There really are only a few _aggressive_ responders that will abrasively tell you to read the docs, but what they really mean is read the code. If that's not appropriate, then simply delete their replies; they really won't mind even a little tiny bit. It's just the
[Asterisk-Users] Re: Cisco SIP license?
In case anybody is trying to work out the currency I used - it's actually British Pounds, but the £ sign isn't being handled by the mailing list. I've noticed that the mailing list is also having problems removing the HTML or Microsoft OLE email components, and is constantly filling the list with the background gunk. It seems to also have a problem with certain platform's line breaks. It's not affected my other emails apart from this last one though... Bizzare... Ad. On 3 Jan 2004, at 5:22 am, [EMAIL PROTECTED] wrote: These optional licenses (which can also be purchase separately, and are=20= approx =A310/$15) are to upgrade the number of users on the Cisco Call=20= Manager Platform. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
On Sat, 2004-01-03 at 06:31, John Coll wrote: SW: Thanks a million for the statement that I only need these two files and they can be just about empty ! David Carter: many thanks for those files which I will study Rich Adamson: That is so re-assuring! That may sound odd but its realy helpful to have the problems I am facing acknowledged and makes me feel that others really see the need for, in effect, intuitive docs to get the novice on-board. I used to write code, now I leave it to my staff, but I guess I can go there. What I am doing is evaluating * to see if we as a company should use and support it rather than just buying in Quintum boxes or whatever. No doubt many others are doing the same. Be aware, that the documentation is getting better. There is info going into the wiki daily and there is a book being written. This kind of documentation is only needed when we get over run by people who aren't willing to take their time learning and willing to spend a lot of effort. We built up a large group of developers and supporters without much documentation. This allowed us to to move forward at a pretty decent pace. Documentation is usually neglected during periods of fast growth. As a company we write software for end-users and I insist that an average 16 year old must be able to make it work, at the basic level, without grief - it must be intuitive. OK make that an average linux administrator for */VOIP but again it really needs to be intuitive - but I guess I am preaching to the convereted. I think you need to go meet and get to know some pbx installers. Maybe make some calls to your local CLECS and requests sales support by a engineer there. You will eventually learn that telephony is a large field that takes quite a bit of effort to understand. Your expectations that telephony be easy will probably not be met unless you have the prerequisite knowledge to begin with. During my last job, I was able to talk to the man who the company hired to install their Intertel pbx. I ended up with the distinct feeling from him that the industry has been progressing much as old trades did. Basically it seemed that one had to study by being a go-fer for a person who knew what they where doing for some time before you could pick up the required knowledge to do simple installs. So while we haven't improved the amount of knowledge you will need to acquire to begin a decent install, but we have developed a community that will help those willing to help themselves. Those who aren't willing to put forth the effort have the ability to pay for the support they need. I would like to offer to try and do that in the wiki - but realistically I don't have the time. Still I am feeling a bit guilty now having got such solid support. The biggest time consumer is the amount you must learn before you can start documenting. There are many people here who are contributing to documentation. What you can do to help now is to ask interesting and directed questions that will be answered by members of this group. When it is sufficiently answered, Olle tends to get it incorporated into the wiki. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Stresstool Help required
Alastair, You were correct. The program was generating the same call IDs for all the INVITEs. In fact it is a small routine of just 2 lines. I checked all my routines, except this one. I never expected a bug in such a small routine. It taught me a good lesson. Sorry, i was not able to post this mail yesterday due to some network problems here. Thanks for helping me out... Warm Regards, Girish From: Alastair Maw [EMAIL PROTECTED] On 02/01/04 14:24, Girish Gopinath wrote: I gave the sip debug command, and one of the lines showed:Ignoring this request Can you log the SIP debug messages to a file and put it up on the web somewhere? Or do an ethereal capture or similar. It's very hard to say what the problem might be without a full SIP trace. It's likely that you're generating the same transaction ID for each SIP INVITE or something silly. Regards, Alastair _ Free transactions in any ATM across India. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Hi Olle! I put something into trouble ticket (I guess you get this as email). BTW 2.03f is available at http://snom.com/download/share. Christian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Olle E. Johansson Gesendet: Donnerstag, 1. Januar 2004 11:57 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone? Christian Stredicke wrote: We at snom have problems with Asterisk when we receive calls without the line indication. When we register we place a contact like this: REGISTER sip:asterisk SIP/2.0 Contact: sip:[EMAIL PROTECTED];line=h35h345 When we receive the 200 Ok, we search for the h35h345. If we dont find it, we try to guess which line is affected. This is relatively easy on a REGISTER response, but on an incoming INVITE we have serious problems. I think some of the challenging and line-assignment problems are related to this problem. Strictly speaking, we register the contact sip:[EMAIL PROTECTED];line=h35h345, NOT sip:[EMAIL PROTECTED]! Parameters are an essential part of a URI which must not be discarded. Ok, Christian, let's fix this. First, I'm curious, is the line= parameter specified somewhere? (Always looking for documentation :-) Secondly, in many places in the sip channel, everything after the ; is discarded. I would really appreciate if Snom could help us fixing this, so Snom phones work correctly and fully with Asterisk. (Have a new Snom 200 on my desk :-) I'm not an experienced C programmer, so I can't fix this myself. However, there are experienced C programmers in the community that will fix this, but they need proper and detailed input on what to fix. I've opened a bug http://bugs.digium.com/bug_view_page.php?bug_id=732 Let's continue adding information there. BTW, there's some other Snom problems where we need input from SNOM. Search on Snom in the bug tracker. Thank you for participating in the Asterisk community! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Stresstool Help required
Alastair, You were correct. The program was generating the same call IDs for all the INVITEs. In fact it is a small routine of just 2 lines. I checked all my routines, except this one. I never expected a bug in such a small routine. It taught me a good lesson. Sorry, i was not able to post this mail yesterday due to some network problems here. Thanks for helping me out... Warm Regards, Girish From: Alastair Maw [EMAIL PROTECTED] On 02/01/04 14:24, Girish Gopinath wrote: I gave the sip debug command, and one of the lines showed:Ignoring this request Can you log the SIP debug messages to a file and put it up on the web somewhere? Or do an ethereal capture or similar. It's very hard to say what the problem might be without a full SIP trace. It's likely that you're generating the same transaction ID for each SIP INVITE or something silly. Regards, Alastair _ Start getting interview calls immediately. http://go.msnserver.com/IN/40245.asp Post your CV on naukri.com today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
Steven - thanks for that. OK I will try and ask interesting and directed questions :) I appreciate the support from several people. Rich Adamson encouraged me to hang in there so I've been back at the shell prompt and edited configuration files down to the bare essentials and still get the same. I would appreciate any suggestions To sumarise: Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and happens to have a firewall connected to the outside but * and the SIP phones are all on the same LAN. The * machine can ping both SIP phones fine. The SIP phones can call each other's IP directly and establish a voice path - but not via * Just in case here is the config of one of the SIP phones Login password xxx MAC 00.0B.82.00.4B.57 IP 10.0.1.202 Subnet 255.255.255.0 (all machines are /24 on this LAN) Default router 10.0.1.198 DNS server #1 10.0.1.198 DNS Server #2 158.152.1.43 SIP Server: 10.0.1.198 Outbound Proxy: SIP User ID:5702 Authenticate ID:5702 Authenticate Password: Name: John Coll 5702 Timezone: GMT SIP User ID is phone number:yes I've experimented with adding an Authenticate Password and adding secret=xxx to sip.conf - but that did not help. Here are the main asterisk configuration files: - ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes - ; ; liza:/etc/asterisk/extensions.conf ; [general] static=yes writeprotect=no ; [globals] CONSOLE=Console/dsp [johnhome] exten = 5702,1,Dial(SIP/5702,20,Ttr) exten = 5702,2,Voicemail(u5702) exten = 5702,102,Voicemail(b5702) exten = 5702,103,Hangup exten = 5703,1,Dial(SIP/5703,20,Ttr) exten = 5703,2,Voicemail(u5703) exten = 5703,102,Voicemail(b5703) exten = 5703,103,Hangup - I have created voicemail boxes for 5702 and 5703 using context johnhome. [EMAIL PROTECTED] johnhome]# pwd /var/spool/asterisk/voicemail/johnhome [EMAIL PROTECTED] johnhome]# l total 16 drwxr-xr-x4 root root 4096 Jan 3 15:41 . drwxr-xr-x4 root root 4096 Jan 3 15:41 .. drwxr-xr-x3 root root 4096 Jan 3 15:41 5702 drwxr-xr-x3 root root 4096 Jan 3 15:41 5703 [EMAIL PROTECTED] johnhome]# - Other configuration files exist in /etc/asterisk [EMAIL PROTECTED] asterisk]# ls adsi.confcdr_pgsql.confjohncoll modem.conf phone.confsamplevpb.conf adtranvofr.conf enum.conf john_todd modules.conf privacy.conf sip.conf z2.conf agents.conf extensions.conf logger.conf musiconhold.conf queues.conf skinny.conf zapata.conf alsa.conffestival.conf manager.conf orig rpt.conf sprackett asterisk.adsiiax.conf meetme.conf oss.conf rtp.conf telcordia-1.adsi asterisk.confindications.conf mgcp.conf parking.conf s2.conf voicemail.conf [EMAIL PROTECTED] asterisk]# - one other that might have some influence perhaps ; ; liza:/etc/asterisk/manager.conf ; [general] enabled = no port = 5038 bindaddr = 0.0.0.0 - I believe that liza:/etc/asterisk/zapata.conf and liza:/etc/zaptel.conf are not relevant but they exist - Starting asterisk up - not too verbose [EMAIL PROTECTED] asterisk]# asterisk -vnc Asterisk CVS-12/12/03-17:13:48, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] [Answer] [BackGround] [Busy] [Congestion] [DigitTimeout] [Goto] [GotoIf] [GotoIfTime] [Hangup] [NoOp] [Prefix] [ResetCDR] [ResponseTimeout] [Ringing] [SayNumber] [SayDigits] [SetAccount] [SetGlobalVar] [SetLanguage] [SetVar] [StripMSD] [Suffix] [Wait] Asterisk Dynamic Loader Starting: [chan_modem.so] = (Generic Voice Modem Driver) = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [res_musiconhold.so] = (Music On Hold Resource) [res_adsi.so] = (ADSI Resource) [res_parking.so] = (Call Parking Resource) [res_crypto.so] = (Cryptographic Digital
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
Hi John, Try adding username=5702 and username=5703 to each of the configs in sip.conf. I recall I had this problem with the Grandstreams. -Original Message- From: John Coll [mailto:[EMAIL PROTECTED] Sent: Saturday, January 03, 2004 11:56 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :) Steven - thanks for that. OK I will try and ask interesting and directed questions :) I appreciate the support from several people. Rich Adamson encouraged me to hang in there so I've been back at the shell prompt and edited configuration files down to the bare essentials and still get the same. I would appreciate any suggestions To sumarise: Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and happens to have a firewall connected to the outside but * and the SIP phones are all on the same LAN. The * machine can ping both SIP phones fine. The SIP phones can call each other's IP directly and establish a voice path - but not via * Just in case here is the config of one of the SIP phones Login password xxx MAC 00.0B.82.00.4B.57 IP 10.0.1.202 Subnet 255.255.255.0 (all machines are /24 on this LAN) Default router 10.0.1.198 DNS server #1 10.0.1.198 DNS Server #2 158.152.1.43 SIP Server: 10.0.1.198 Outbound Proxy: SIP User ID:5702 Authenticate ID:5702 Authenticate Password: Name: John Coll 5702 Timezone: GMT SIP User ID is phone number:yes I've experimented with adding an Authenticate Password and adding secret=xxx to sip.conf - but that did not help. Here are the main asterisk configuration files: - ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes - ; ; liza:/etc/asterisk/extensions.conf ; [general] static=yes writeprotect=no ; [globals] CONSOLE=Console/dsp [johnhome] exten = 5702,1,Dial(SIP/5702,20,Ttr) exten = 5702,2,Voicemail(u5702) exten = 5702,102,Voicemail(b5702) exten = 5702,103,Hangup exten = 5703,1,Dial(SIP/5703,20,Ttr) exten = 5703,2,Voicemail(u5703) exten = 5703,102,Voicemail(b5703) exten = 5703,103,Hangup - I have created voicemail boxes for 5702 and 5703 using context johnhome. [EMAIL PROTECTED] johnhome]# pwd /var/spool/asterisk/voicemail/johnhome [EMAIL PROTECTED] johnhome]# l total 16 drwxr-xr-x4 root root 4096 Jan 3 15:41 . drwxr-xr-x4 root root 4096 Jan 3 15:41 .. drwxr-xr-x3 root root 4096 Jan 3 15:41 5702 drwxr-xr-x3 root root 4096 Jan 3 15:41 5703 [EMAIL PROTECTED] johnhome]# - Other configuration files exist in /etc/asterisk [EMAIL PROTECTED] asterisk]# ls adsi.confcdr_pgsql.confjohncoll modem.conf phone.confsamplevpb.conf adtranvofr.conf enum.conf john_todd modules.conf privacy.conf sip.conf z2.conf agents.conf extensions.conf logger.conf musiconhold.conf queues.conf skinny.conf zapata.conf alsa.conffestival.conf manager.conf orig rpt.conf sprackett asterisk.adsiiax.conf meetme.conf oss.conf rtp.conf telcordia-1.adsi asterisk.confindications.conf mgcp.conf parking.conf s2.conf voicemail.conf [EMAIL PROTECTED] asterisk]# - one other that might have some influence perhaps ; ; liza:/etc/asterisk/manager.conf ; [general] enabled = no port = 5038 bindaddr = 0.0.0.0 - I believe that liza:/etc/asterisk/zapata.conf and liza:/etc/zaptel.conf are not relevant but they exist - Starting asterisk up - not too verbose [EMAIL PROTECTED] asterisk]# asterisk -vnc Asterisk CVS-12/12/03-17:13:48, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] [Answer] [BackGround] [Busy] [Congestion] [DigitTimeout] [Goto] [GotoIf] [GotoIfTime] [Hangup] [NoOp] [Prefix] [ResetCDR] [ResponseTimeout] [Ringing] [SayNumber] [SayDigits] [SetAccount] [SetGlobalVar]
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
On Sat, 2004-01-03 at 17:55, John Coll wrote: ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes And now dial 5703 from 5702. 5703 rings but when 5703 is taken off hook no voice path is established and both phones give rapid beep beep beep after a few seconds. The following has been cut a bit but I hope I've left something useful in there Isn't this the codecs problem? Try adding :- disallow=all allow=ulaw allow=alaw into both of your sip.conf descriptions. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
John, Obviousely, this would not work. Look at my example before; [5702] == type=friend username=5702 == context=internal dtmfmode=info username and context should match. Better get it working in a simple LAN first, why NAT, why voicemail .. Go to basics :), SW From: John Coll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :) Date: Sat, 3 Jan 2004 16:55:45 - Reply-To: [EMAIL PROTECTED] ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
Hi! You started out with a much too complex setup. Start small, test, and then add things step by step - don't configure everything at once! Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and happens to have a firewall connected to the outside but * and the SIP phones are all on the same LAN. Then remove the nat=yes statements (and also the reinvite= and canreinvite= parts) - for Asterisk your phones are not network address translated. Also I guess you already read somewhere that having Asterisk behind NAT is cause of trouble - but that doesn't matter for your internal testing with a basic setup. After adjusting sip.conf don't forget to reload Asterisk. externip = 10.0.1.198 Remove this for the time being to get your internal test running. Leave messing with externip= for later. [5702] type=friend host=dynamic context=johnhome qualify=yes callerid=John workroom #1 5702 mailbox=5702 disallow=all allow=ulaw allow=alaw [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 disallow=all allow=ulaw allow=alaw Found audio format UNKN Found description format PCMU Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 See the disallow and allow statements I added in sip.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
Dave You were right! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton Sent: 03 January 2004 17:19 To: Asterisk List Subject: RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :) On Sat, 2004-01-03 at 17:55, John Coll wrote: ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes And now dial 5703 from 5702. 5703 rings but when 5703 is taken off hook no voice path is established and both phones give rapid beep beep beep after a few seconds. The following has been cut a bit but I hope I've left something useful in there Isn't this the codecs problem? Try adding :- disallow=all allow=ulaw allow=alaw into both of your sip.conf descriptions. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
On Sat, 2004-01-03 at 18:59, John Coll wrote: Dave You were right! In the words of that welsh comedian I know because I was there. As others have said there's a steep learning curve for *, but as one who's climbed just some of it, I can say it's worth it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
Dave You were right disallow=all allow=ulaw allow=alaw gave me two-way voice! Whew! Thanks a million. I wonder if I really should have found that for myself ... I've added it to the voip-info.org wiki OK lets see if the next step is a bit easier :) thanks again all john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton Sent: 03 January 2004 17:19 To: Asterisk List Subject: RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :) On Sat, 2004-01-03 at 17:55, John Coll wrote: ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes And now dial 5703 from 5702. 5703 rings but when 5703 is taken off hook no voice path is established and both phones give rapid beep beep beep after a few seconds. The following has been cut a bit but I hope I've left something useful in there Isn't this the codecs problem? Try adding :- disallow=all allow=ulaw allow=alaw into both of your sip.conf descriptions. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP/grandstream not registering
And why you have two different entries for the same object? Posting two times the same questions with other data will not help to resolve the issue more quickly... - Original Message - From: Glenn Dalgliesh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 4:00 AM Subject: Re: [Asterisk-Users] SIP/grandstream not registering It looks like you have you * on public IP and your phones on private, most likely behind NAT if so in your sip.conf under each [grandstreamX] you most likely need: nat=yes - Original Message - From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 1:44 AM Subject: [Asterisk-Users] SIP/grandstream not registering hi, i can't seem to register my grandstream SIP to * server... i have my grandstream IP as 192.168.0.11 want to register to * at 202.51.xx.xxx. sip show peers says that my grand stream has unspecified IP but when i try to dial a number it gets this error... WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 40939 (Response) my sip.conf is... [grandstream2] type=peer host=dynamic secret=grandstream2 reinvite=no canreinvite=no qualify=60 [grandstream2] type=user host=dynamic secret=grandstream2 context=outgoing reinvite=no canreinvite=no qualify=60 help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
Add this to ur sip.conf ..that would help u. disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm -B And sip.conf contains this [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 10.0.1.198 ; Addres [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=no [5703] is similar extensions.conf is currently slightly modified verson of Zac Sprackett's file http://sprackett.com/asterisk/ - its a bit long so I won't paste yet. However I have had the same result with a much simpler extensions.conf - some days ago. Any help would really be appreciated as I am stuck and finding the process hard because I can't seem to find a basic introduction aimed at getting me up and running with the most basic of systems. Perhaps you can point me to a BASIC and minimal set of configuration files for example for a SIP phone or two on a NAT LAN with an X100P plugged into PSTN. I guess that is where most people start - or should I start somewhere else? I've been at this off and on for two weeks Linux admin and firewalls seem trivial compared to this so I must be missing something pretty basic :) thanks john ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] expression parsing
Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable
is not defined, I will get a parse error. Yeah, there are ways around it, but I would
think that it should return false if 0, null, or undefined. I would change it, but I
have no idea about bison and I only have very basic C skills.
There was a bug opened on this, and there was a valid work-around posted, but I would
think that it would be 'nicer' if it would evaluate it this way. (Ref:
http://bugs.digium.com/bug_view_page.php?bug_id=401 ) If you put a 0 after the }
it does work as I would want it to without an error. The other suggestions did not
work. I propose for this bug to be re-opened.
extensions.conf:
exten = 1234,3,GotoIf($[${a}]?4:5)
If a is undefined:
WARNING[37910]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror(): syntax error:
parse error
-- Executing GotoIf(SIP/1240-5eb6, 0?4:5) in new stack
-- Goto (default,1234,5)
If I change the extention to exten = 1234,3,GotoIf($[${a}0]?4:5) it works as expected.
Also, I'm not sure if this is my bad or what. If I use exten =
1234,3,GotoIf(${a}?4:5) and a is undefined:
-- Executing GotoIf(SIP/1240-2b8a, ?4:5) in new stack
-- Goto (default,1234,4)
it still returns true.
Behold the power of *,
Kevin
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RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? Send via SIP, RTP or INLINE AUDIO. Make sure you change your dtmfmode= in your sip.conf to match the mode set on the phone.. Yes.. that solved it. I added dtmfmode=info to sip.conf and set SIP INFO as the DTMF type on the phone and all is now well! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Ah. I suppose this isn't you, either. http://www.worldogl.com/view_clan_info.php?clanid=5363 On Saturday 03 January 2004 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free PSTN calls
I have set up my * box to provide free calling. You can access it by dialing 1-700-945-2475. Once you hear the prompt dial 1 the area code and number. I would also like to test some direct incoming IAX connections from some other * boxes to see if I can terminate PSTN calls that way. If you would like to help me testemail me: [EMAIL PROTECTED] Also, more information is available at http://www.freephoneproject.com/nexthop Isaac [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
That was the beginning of the all female quake clan girlz of destruction http://www.girlzgotgame.net/. Annie and I joined a 2v2 ladder. Yes, that's me, however, that was nearly 3 years ago. Why bring this up? What does this have to do with our discussion? Are you just trying to prove your skills with Google? Go do it in IRC, not this forum please. Barbara Simpson Qwest Voice Over Packet Services --- Tilghman Lesher [EMAIL PROTECTED] wrote: Ah. I suppose this isn't you, either. http://www.worldogl.com/view_clan_info.php?clanid=5363 On Saturday 03 January 2004 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users === message truncated === __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
On Saturday 03 January 2004 18:37, Me wrote: That was the beginning of the all female quake clan girlz of destruction http://www.girlzgotgame.net/. Annie and I joined a 2v2 ladder. Yes, that's me, however, that was nearly 3 years ago. Why bring this up? What does this have to do with our discussion? Are you just trying to prove your skills with Google? Go do it in IRC, not this forum please. Barbara Simpson Qwest Voice Over Packet Services So why are you here, again, since you've already indicated how bad you think Asterisk is? Perhaps Qwest has a competing service and wishes to monitor its competitors? Why else would you want to drive newbies away, yet stick around to continue to watch (and post to) the list? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mini-ITX suggestions
We just got 1 Nehemiah in the office. Performance-wise it's pretty much a P3-class machine, IIRC the FPU is running at full clock speed compared to the 800MHz version. We do have problem booting a 686 optimized kernel on it. Can't install White Box Enterprise Linux (community distro based on RedHat Advance Server 3.0) but RH 8 works fine. The box we have takes 2 PCI cards beautifully, but no * yet. I'd * running on the 800MHz box. It's works fine as long as you're not using too much floating point (i.e. avoid compression/decompression on the box). My setup had one XP101 connected to one 1 DG104S MGCP gateway and one BATM MGCP gateway. Works pretty well without MOH (MP3=FPU). MeetMe works fine. I'd an early TDM400 hooked up to it for a while, but gave up because of the massive amount of PSU noise. YMMV. FYI. Patrick Cantwell wrote: On Fri, 2 Jan 2004, Steven Critchfield wrote: On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote: Does anyone have recommendations for (or against) mini-ITX platforms to be used with Wildcard X100P and TDM400P cards? I am considering the use of systems using VIA EPIA CL and Epia M as small, quiet platforms on which to host Asterisk. It has been covered here before that those are i586 level chips, so be prepared for the actual horsepower you will get out of them. If you don't do much VoIP, it will probably be fine. I think there is even a person or 2 here already using them as thats why we know about the i586 problem. What about the newer chip (the Nehemiah)? I have an 800mhz version of the older system/core that's clearly i586 architecture doing some PVR stuff (I have a tuner card with mpeg encoding builtin, so the lack of horsepower is a non-issue), however I *believe* the newer Nehemiah core is i686 compatable. I do know they've added hardware RNG, SSE/SIMD extensions, and a full speed floating point unit to the processor. The motherboard also has an updated north/south bridge, utilizing PC2100/DDR266 RAM, comes with USB2.0 and firewire integrated, and moves from 1 ATA100 to 2 ATA133 channels. It'd sure be a waste to couple all of that nice hardware to an antiquated i586 chip. (and the best is a board+chip+case can be accomplished for around $200USD) Can anyone confirm if they've added what was necessary to bring the chip up to the i686 level? A brief overview of the new Nehemiah features can be seen at http://www.mini-itx.com/reviews/nehemiah/?page=3#s05 I'll have one of these machines in hand probably early next week, so maybe I can do some testing before I move it into the role of my new PVR (those damn trans-encodes from mpeg to streamable formats take too darn long on the 800! :) -Pat -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Re: Cisco SIP license?
The £ came through here OK... --- These optional licenses (which can also be purchase separately, and are approx £10/$15) are to upgrade the number of users on the Cisco Call Manager Platform. --- Dan (in UK) On Sat, 2004-01-03 at 13:34, Adthrawn wrote: In case anybody is trying to work out the currency I used - it's actually British Pounds, but the £ sign isn't being handled by the mailing list. I've noticed that the mailing list is also having problems removing the HTML or Microsoft OLE email components, and is constantly filling the list with the background gunk. It seems to also have a problem with certain platform's line breaks. It's not affected my other emails apart from this last one though... Bizzare... Ad. On 3 Jan 2004, at 5:22 am, [EMAIL PROTECTED] wrote: These optional licenses (which can also be purchase separately, and are=20= approx =A310/$15) are to upgrade the number of users on the Cisco Call=20= Manager Platform. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
As far as your original post goes, Asterisk doesn't regularly segment fault. There are many stable installations. We have a bunch of happy users. This is remarkable since Asterisk is still a beta product. There is plenty of useful information on the sites you panned if you are smart enough to find it. Paid support is available from Digium if you don't like the free support or are to dumb or lazy to figure things out for yourself. The Asterisk community is in general very supportive of beginners, although not of stupidity or sloth. Lastly, anything that may be wrong with Asterisk is being fixed in a big hurry. It's already better than the proprietary systems, even though it's still a beta product, and it's free for god's sake. In a few years, Asterisk will be a polished product in general release and you will have missed the boat. Don't quit your day job. It's not going to be easy to find someone else who will hire you. You should find a different group to read. Regards, Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, January 03, 2004 5:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms.
Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something useful pl. -B - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 5:36 PM Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk.
[Asterisk-Users] New to asterisk? RUN... don't walk.
I now stand by my original post with more conviction than ever. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. How does the saying go? If you're not part of the solution, As one who is new to Asterisk, I am happy that I did not heed your suggestion to RUN. The sources of information you deem worthless have proven quite helpful to me in (so far) setting up my Asterisk server, getting SIP phones talking through it, using voicemail, and amazingly- all this without a single segmentation fault! :-P Your complaints about the Asterisk Community remind me very much of complaints often made about the Linux Community. Judging an entire community (and even quality of the software) based on the actions of a few people is a big mistake. -Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Paul Mahler wrote: As far as your original post goes, Asterisk doesn't regularly segment fault. There are many stable installations. We have a bunch of happy users. This is remarkable since Asterisk is still a beta product. 888.480.4638, my toll-free number, is routed to wherever I choose to route it to via an Asterisk box. (Usually it rings through to my cell phone.) There is the occasional hiccup, but in general I have to throw in a Me too in reply to Paul. The software is generally quite stable. Having played with an Asterisk installation and read about the things that are going on here, I am constantly amazed at all the things it's being made to do. -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Geek In Charge * 888.480.4NET (4638) * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P driver modprobe failed
Hello Everyone, I just got my Dev Kit TDM today... :D I installed the X100P ok (wcfxo); however, when I tried to 'modprobe wcfxs' for the TDM400P(TDM10B), I got this error message: /lib/modules/2.4.19/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters /lib/modules/2.4.19/misc/wcfxs.o: insmod /lib/modules/2.4.19/misc/wcfxs.o failed /lib/modules/2.4.19/misc/wcfxs.o: insmod wcfxs failed I checked the TDM10B card installation; I checked the 12V power connecting to it; I moved the card to a different PCI slot...but all of that didn't help the driver installation :( Help? Any suggestion would be appreciated! Thanks, Michael W. p.s. I compiled the driver from zaptel-0.7.0.tar.gz. - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java?
On Thu, 01 Jan 2004 17:50:32 +0100 Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) That's an excellent suggestion, I agree with Ray. Masakazu, do you think you could provide a working sample either here on the list or in the Wiki? Cheers, Philipp yeah. surely ok. but please just a moment to disclose my code. because that is very evalution code at now. bit buggy ;-) I think AMP + ming + actionscript + * + ecasound + xoops makes us good CRM ( one of anti-claimer ) enviroment. http://www.wakkanet.fi/~kaiv/ecasound the interface of voice by asterisk. and translate to mp3 by ecasound. and play that realtime stream by ming. mack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
Tilghman, Thanks for bringing this to my attention. I agree with the comment from 'siggi' - it seems that this should be configurable on a per handset basis, not per voicemail user as currently implemented in the most recent patch that has been hung on this bug. It's a useful hack that may help us in the short term and I'm grateful for the pointer, but doesn't seem like anything that should ever find its way into CVS. It's not clear to me who authored this patch (Corydon76 ), or why 'bkw918' closed it at the author's request. Without support for personal names and signature blocks this bug tracking software is awfully dehumanizing, and I'm finding myself a little more disoriented that normal ;-) -d -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 02, 2004 2:27 AM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? On Thursday 01 January 2004 12:57, Darren Nickerson wrote: That worked a treat - thanks! Comedian Mail is now able to download to the handset and there's a lot more functionality now. There's a patch on the bugtracker that should allow you to specify these codes per user, as requested. http://bugs.digium.com/bug_view_page.php?bug_id=733 Let me know how it works for you. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Residential router w/ QoS support?
Thilo, I wasn't too sure about the packet based prioritization so I stuck with the physical port based model. That is, I made port 1 high priority all the time, then plugged my * sever into that port. Actually, that segment has all the ip phones and nothing else. The problem I had initally with my Vonage installation (now defunct in favour of VocePulse Connect) was that FTP'ing large amounts of data would kill voice line separate segments, and assigning the VOIP segment priority took care of it entirely. I don't have so many devices around that making the separate segment was an issue. Michael On Fri, 02 Jan 2004 18:26:57 +0100, Thilo Salmon wrote: Michael, I just got mine. Do you recall how you managed to priortize RTP? Or do you rely on the 'priortized switching port' feature? I tried that, but perhaps my TOS value does not match the one this router expects. Even sending a single, large email can kill the voice stream. Leave alone BitTorrent. Unfortunately, the documentation is far from extensive... Thilo On Thu, 2003-12-18 at 17:15, Michael Graves wrote: I use a Linksys BEFSR81 which ia an 8 port model with QoS. I paid about $90 USD. I had to buy a QoS router when I first installed a Vonage line about a year ago. Without it using FTP to d/l loarge files would simply kill my calling. Michael On Wed, 17 Dec 2003 17:01:52 +0100, Thilo Salmon wrote: Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 Philosophers and plowmen, each must know their part to sow a new mentality, closer to the heart. - Geddy Lee, Rush ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- [netzquadrat] GmbH | Thilo Salmon Ronsdorfer Str. 74 | Fon: +49 211 302033 12 40233 Duesseldorf| Fax: +49 211 302033 22 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 Philosophers and plowmen, each must know their part to sow a new mentality, closer to the heart. - Geddy Lee, Rush ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ===END FORWARDED MESSAGE=== -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 It'll all go back to normal if we put our nation first. But the trouble with normal is it always gets worse. - Bruce Cockburn ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call instances? That is, would the snom 200 be extension 2000 or 2000-2004? Also, did the 2.03f firmware resolve the matter? Thanks, Michael On Sat, 3 Jan 2004 16:05:39 +0100, Christian Stredicke wrote: Hi Olle! I put something into trouble ticket (I guess you get this as email). BTW 2.03f is available at http://snom.com/download/share. Christian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Olle E. Johansson Gesendet: Donnerstag, 1. Januar 2004 11:57 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone? Christian Stredicke wrote: We at snom have problems with Asterisk when we receive calls without the line indication. When we register we place a contact like this: REGISTER sip:asterisk SIP/2.0 Contact: sip:[EMAIL PROTECTED];line=h35h345 When we receive the 200 Ok, we search for the h35h345. If we dont find it, we try to guess which line is affected. This is relatively easy on a REGISTER response, but on an incoming INVITE we have serious problems. I think some of the challenging and line-assignment problems are related to this problem. Strictly speaking, we register the contact sip:[EMAIL PROTECTED];line=h35h345, NOT sip:[EMAIL PROTECTED]! Parameters are an essential part of a URI which must not be discarded. Ok, Christian, let's fix this. First, I'm curious, is the line= parameter specified somewhere? (Always looking for documentation :-) Secondly, in many places in the sip channel, everything after the ; is discarded. I would really appreciate if Snom could help us fixing this, so Snom phones work correctly and fully with Asterisk. (Have a new Snom 200 on my desk :-) I'm not an experienced C programmer, so I can't fix this myself. However, there are experienced C programmers in the community that will fix this, but they need proper and detailed input on what to fix. I've opened a bug http://bugs.digium.com/bug_view_page.php?bug_id=732 Let's continue adding information there. BTW, there's some other Snom problems where we need input from SNOM. Search on Snom in the bug tracker. Thank you for participating in the Asterisk community! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I am easily satisfied with the very best - Winston Churchill The questions arisen, is this a prison? Some say it is, but I say it isn't. - Ian Hunter ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ===END FORWARDED MESSAGE=== -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I know nothing, but I keep listening. - INXS ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
On Saturday 03 January 2004 22:51, Darren Nickerson wrote: On Friday, January 02, 2004. Tilghman Lesher wrote: On Thursday 01 January 2004 12:57, Darren Nickerson wrote: That worked a treat - thanks! Comedian Mail is now able to download to the handset and there's a lot more functionality now. There's a patch on the bugtracker that should allow you to specify these codes per user, as requested. http://bugs.digium.com/bug_view_page.php?bug_id=733 Let me know how it works for you. Thanks for bringing this to my attention. I agree with the comment from 'siggi' - it seems that this should be configurable on a per handset basis, not per voicemail user as currently implemented in the most recent patch that has been hung on this bug. It's a useful hack that may help us in the short term and I'm grateful for the pointer, but doesn't seem like anything that should ever find its way into CVS. It's not clear to me who authored this patch (Corydon76 ), or why 'bkw918' closed it at the author's request. Without support for personal names and signature blocks this bug tracking software is awfully dehumanizing, and I'm finding myself a little more disoriented that normal ;-) I authored the patch, and I had it closed because it although it was a method to customize it, as siggi noted, it's not the right way. I thought about modifying the patch, but I don't know enough about ADSI to make intelligent choices about coding it in a handset specific way. It'd be coding in the dark, and I don't like doing that. What I had done was based upon your post to the list. If you'd like to flesh out exactly how it needs to be done (i.e. the ADSI terminology), I can look at doing it the right way. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
