Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
For another tone frequency for the outside dialtone, try putting this value [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED];*(.4/0/1),10(*/0/2+3) in the Outside Dialtone field. It will give you a slight pause followed by a different dialtone frequency. On a Linksys/Siprua 941, that would be at the top of the Regional page. However, you won't hear any secondary dialtone unless you put a comma after EVERY initial '9' in the dialplan string for each line in use. On a 941, that would be at the bottom of the Ext 1 and Ext 2 pages of the web interface. I suggest the dialplan string of: (*xx|[1-7]xx|9,[3469]11|98|99|9,[2-9]xx|9,11|9,[2-9]xx|9,1[2-9]xx[2-9]xx|9,011xxx.) - Walt Joyce Eric ManxPower Wieling wrote: I can't help you with that. I only wanted to point out that ignoreopat is not what you need. On Polycom SIP phones you continue dialtone by placing a , in the phone's dialplan. SIP phones have their own internal dialplan that is not part of Asterisk's dialplan. You would have to check the docs for your phone. Not all SIP phones can continue dialtone. bilal ghayyad wrote: I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone, not a dialtone provided by Asterisk. Al lists wrote: Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Panel?
Matt - I'd like the sourcecode for the SIP panel. - Walt Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Terry Giufre-Sweetser wrote: Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? I wrote a program a while ago - don't know if it will still work: http://www.sineapps.com/sinepeers.php Let me know if you want the sourcecode, it's probably buried somewhere in my svn repository. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAdqJDQNt8rg0Kp4RAjbCAKCt01nH1hGq3estWpoFLeYsdypq6QCgvaHf Eeo66dSiiOKJmkqoohsoT8g= =rcag -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Panel?
Yes, I have. It is not difficult. I use the Asterisk Manager interface. Is there a particular question? - Walt Terry Giufre-Sweetser wrote: Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] surge protector?
On Sun, Jul 15, 2007 at 10:36:19AM -0400, Robert Moskowitz said: Joe acquisto wrote: APC makes a two line unit. PTEL2. But it's two lines in one jack. I have always been a fan of Triplite. They use old tech when appropriate. I am big on Line Conditioners and UPSs with line conditioning. Of course power in my house is really bad. Anything big kicks on and lights flicker (high risers off of sags are worst, and much more common that spikes above norm). Of course you need Telco protection, not AC protection (well you need both). The Triplite gear that I have looked at had very simple Thiristor (sp?) dumps to ground and fast clamping. Another - www.ablecom.com is a bit more Pro Just do a google and take your pick. joe a. On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote: I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd I always found that the stuff built-in to power strips and UPS's a little lacking, and prefer products designed for telecom to begin with: http://www.sandman.com/surge.html http://www.telephonecentral.com/category.aspx?categoryID=65 http://www.citelprotection.com/citel/telephone.htm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
On Wed, Jul 11, 2007 at 09:34:51AM -0600, Stephen Bosch said: Walt Reed wrote: No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. I don't buy this. This problem did not appear until Digium changed list servers. So you don't believe that the new servers may have shorter timeouts than the old and that recipient configuration has no impact? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. Many people fail to realize that high-volume mail servers (especially for large mailing lists) don't have long timeouts and therefore can't tolerate slow recipient servers. It takes too many resources. Make sure that you whitelist list mail at all phases of your protection systems. Make sure you are NOT doing sender callouts, running every message through spamassassin, greylistging, etc. for list mail. Lastly, there is nothing Digium is going to be able to do if your DNS servers are flakey, or route path is. Headers just tell you that there is a delay. We already know this. Only the sending AND receiving server logs can tell you WHY, and then you may only know if the session was run in debugging mode. On Wed, Jul 11, 2007 at 07:33:43AM -0400, Jared Smith said: On 7/11/07, Bill Maidment [EMAIL PROTECTED] wrote: email delays here are about 8 days. I don't expect to see this until 19th July When you do get the message, please reply with the email headers, so we have some chance of tracking down the problem. For example, below ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
No, please don't post headers. Headers tell us nothing. They don't tell us such things such as DNS resolution problems, routing problems, if the recipient's server is tempfailing, etc. The ONLY thing really useful are the mail logs from the list server. The only people that have access are Digium employees. I encourage everyone who is having delays to visit http://lists.digium.com/mailman/listinfo/asterisk-users and look at the bottom of the page. The link to [EMAIL PROTECTED] and the other two addresses listed are the ONLY addresses that should be notified of list problems / delivery issues. Considering that so Very Very few subscribers are having delays, there is a 99% chance that you have something messed up on your side - DNS reliability, your network, one or more of your MX servers, some goofy anti-spam scheme, etc. In this case: dig mx mailcall.com.au ;; ANSWER SECTION: mailcall.com.au.60 IN MX 100 mx2.zoneedit.com. mailcall.com.au.60 IN MX 0 email.mailcall.com.au. Connection attempts to mx2.zoneedit.com were taking well over a minute to get the 220 mx2.zoneedit.com ESMTP Postfix response. Most high-volume list servers won't wait that long. I strongly suggest you find a better backup mail relay service, or don't even list a second MX. Once ALL your MX servers (no matter what priority they are listed at) are working quickly and correctly, THEN contact the list admins for further help. On Tue, Jul 10, 2007 at 12:18:32AM -0600, Anthony Francis said: Most of the users using this list do not experience the issue you are having, rather than insult the admins, please trouble shoot and if you cannot, at least post headers so others can. -- Original Message -- From: Dimitri Volski [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Tue, 10 Jul 2007 11:32:49 +1000 There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! ;) -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow list
On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said: Well, this is now the third active thread on this subject, but I guess you won't see this message for a while. Has anyone dissected the headers of a delayed message yet? We should be able to tell for sure where the holdup is. All of the messages are coming through on time for me, so it won't do much good for me to look. Looks like mail is getting held up between INXS.digium.internal and lists.digium.com INXS.digium.internal received it the first of July, lists.digium.com received it on the 4th. drdos.info (ME) received it from lists.digium.com on that same day (Today). What you can't see without looking at the mail server logs on both ends is delivery attempts. Greylisting for example can totally hose you over depending on the implementation. Greylisting without whitelisting is irresponsible. How many tries did the digium server make before the message finally got through??? That's what we need to know. Only Digium can say. Before poking Digium too much, I would look at exactly what YOUR mail servers are doing that may potentially be the real cause of the delays. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk
On Tue, Apr 10, 2007 at 10:28:46AM -0400, Mike said: Probably, if I only needed one FXO. What is the customer has 4 channels (PSTN lines)? Don't I need 4 FXO? And, about the Sipura, it looks like it would do what I want, but it only has one FXO, limiting it's usefulness. I strongly recommend that you check out the wiki: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell poweredge 860 acceptable for office environment ?
On Fri, Mar 16, 2007 at 01:08:54PM +0100, Olivier said: I'm really after 1U-2U silent servers as I've got the feeling most of them are too noisy for offices and most of our clients don't have server rooms. Herin lies the problem. Slimline rackmount servers require several small fans operating at high speeds to remove the heat. Any fan at high RPM is going to be noisy. You are better off going with a larger box like a 4 or 5U unit with a couple larger PWM fans than can be slowed down and therefor be much more quiet. Seagate hard drives (SATA) also tend to be more quiet than server class scsi drives, or other brands. In a server, it's tough to get silent, but you can sure get a lot better than average. Servers in general tend to be noisier because they tend to be in closets or server rooms. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
On Thu, Mar 15, 2007 at 08:08:57AM -0600, Joe Greco said: Anyone who's been in the industry for any length of time will have stories. Some of them even interesting. I remember a few years ago when the roof/wall of an ATT data center was destroyed during a storm. Yep. Ashburn VA datacenter. Tornado hit it. Water was pouring in on someone's servers, and surprisingly they didn't go down! ATT did bring them down due to safety concerns. Our servers, in that datacenter, were unaffected and had zero downtime. Most of the damage was to unoccupied portions of the datacenter. I've also had multiple drives fail simultaineously on a 0+1 Raid. It totally sucks when it happens. One online spare was not enough and didn't have time to rebuild before the second and third drives failed. We did have backups and were able to restore everything within 4 hours, but we still lost some data between the last backup the night before and when the drives failed. These were not cheapo IDE drives either, they were server grade scsi (HP branded Seagates.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List problem handling HTML E-mails?
On Wed, Feb 07, 2007 at 11:45:30AM -0800, Yuan Liu said: My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing mail and checked that it's correct. Anyone else experiencing same? It's bad netiquette to send HTML to mailing lists in general. It hoses up digests and archives, and some people don't have HTML capable clients. Some mail clients send both plain text and html, which isn't quite so bad since the receipient / archiving software can pick out the plain text version, but clients that send HTML ONLY should be avoided. Check to see if you can configure your mail service to use plain text and that should fix things. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
On Wed, Jan 03, 2007 at 04:51:23PM -0600, John French said: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. While I can't answer that particular question, have you considered passive noise reduction techniques such as reorientation / relocation? Sometimes just turning a component 90 degrees (horizontally or vertically) can reduce the perceived noise... Then there is the option of using some kind of enclosure... Usually enclosures can reduce noise greatly while not significantly impacting cooling (such as a wall-mount cabinet.) Also, the use of noise absorbing materials can help. (google for acoustical panels and acoustical foam) Obviously, locating equipment like this in a quiet workspace is not ideal... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.0, IMAP and Dovecot
As an FYI, this has absolutely NOTHING AT ALL to do with Asterisk.
Asterisk is a PBX application. Dovcot is an email application. They have
nothing to do with each other. Asterisk is not a Linux distribution or
operating system.
I suggest that you ask your question on a more appropriate mailing list,
but it seems as though you are so thourougly confused as to what is
what, that you should probably pick up a beginners book on Linux and go
from there. I know this may sound harsh, but trust me: you will be much
better off in the long run if you educate yourself somewhat.
To give you an idea on how far off you are, it would be like going into
a car repair shop and asking about a furnace problem. While someone
there may be able to help you, you went to the wrong place.
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin said:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Walt Reed
[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.0, IMAP and Dovecot
Dan, Please accept my sincerest appology. I had my head thoroughly up my
back orifice. I haven't kept up with the new IMAP feature in 1.4
I'll go back in my corner now :-)
On Wed, Dec 27, 2006 at 02:19:07PM -0500, Walt Reed said:
As an FYI, this has absolutely NOTHING AT ALL to do with Asterisk.
Asterisk is a PBX application. Dovcot is an email application. They have
nothing to do with each other. Asterisk is not a Linux distribution or
operating system.
I suggest that you ask your question on a more appropriate mailing list,
but it seems as though you are so thourougly confused as to what is
what, that you should probably pick up a beginners book on Linux and go
from there. I know this may sound harsh, but trust me: you will be much
better off in the long run if you educate yourself somewhat.
To give you an idea on how far off you are, it would be like going into
a car repair shop and asking about a furnace problem. While someone
there may be able to help you, you went to the wrong place.
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin said:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Walt Reed
[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
--
Walt Reed
[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 Second Incoming Call
On Wed, Nov 29, 2006 at 11:34:41PM +0200, Dovid B said: I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times dosent realize that a new call is coming in. Thanks. I can't offer any help here, but just a ditto to your question. Nothing seems obvious to me that would change this behavior in the XML. The problem is annoying enough that I was thinking of writing a little desktop applet that would popup with this info, but the phone should do this by default. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manage Users in LDAP
I also use ldapadmin, but for many common tasks, I use custom command line scripts that wrap standard ldap commands. I also wrote a couple simple CGI's that allow users to change their password, select their preffered shell, update GECOS, and a few other options. If you are managing Asterisk users in LDAP too, I would imagine that a custom CGI for managing asterisk specific attributes would be very useful. Yes, templates within ldapadmin can also work, but sometimes you want something more appropriate for end-users. On Tue, Nov 28, 2006 at 09:29:50AM -0600, Ejay Hire said: I second the vote for ldapadmin. You can extend it with custom templates for your asterisk specific attributes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Clock Signal Problem
I am using a Digium TE110P to connect to my local telco's PRI network. The problem is that I do not pick up a clock signal from the telco. According to zttool and /proc/zaptel/1 the sync source is 'internally clocked'. By not using the telco's clock source I'm having problems with faxes and occasional HDLC errors and dropped calls. /etc/zaptel.conf looks like this: span=1,1,0,ccs,hdb3 dchan=16 bchan=1-15,17-31 I suspected the PRI line may be faulty so I moved the server to another location where I know I'm picking up a clock source on a 4 port Digium card. The problem persisted on the single port TE110P. I then replaced the TE110P with an identical TE110P and the new card still uses the internal clock. I even tried 3 different motherboards (2 Intel and 1 Gigabyte) but nothing changed. Did anyone have a similar problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729a.so coredump in SVN trunk
On Tue, Aug 08, 2006 at 12:08:43PM +0100, [EMAIL PROTECTED] said: Hi i just setup asterisk and tried to install the digium g729 codec. it works ok using stable but with SVN i get a core dump with the error, 'missing mod_data for codec_g729a.so' i had a quick look though the archives but all i came up with is this, http://threebit.net/mail-archive/asterisk-dev/msg02414.html as this is 3 months old i was wondering if there has been any progress on this? and waht the plans are for this g729 codec? will i have to wait to 1.4 before i can use the cool features in the current trunk and g729 or is there a 'dev' version of teh g729 module that works with SVN? As the OP of that message, I'm still waiting too. I believe that trunk is in a feature freeze at this point, so the module interfaces should be stable now. It would be very nice to have this module updated - it would allow more people to run the new code and flush out bugs prior to final release. It's a win-win. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk BRI in the USA
On Wed, Apr 12, 2006 at 11:16:10PM -1000, Mark Coccimiglio said: I'm seeing Diva Server V-BRI running close to $1K/card. There are other Diva cards running around $700. A little pricy but not impossible to do. I remember back in the 90's I had ISDN into my home for internet access. The netgear router I used cost me about $350 back then, and it worked great. I still have it as a matter of fact. However internet access is not what I need. I'm still waiting for the ILEC (HawaiianTelcom) to get back to me to find out if it is even possible to do BRI into my office. The nearest ISDN capable CO is located a bit of a distance from my office (actually its closer to my home). The local CO dosen't have BRI capablities. From what I'm hearing when you bundle together all the costs BRI PRI are gonna be close in price (from a H/W point of view.) Maybe I should just look into going the PRI route and try to find some people willing to buy on my extra DiD's? Any one what a phone number in Hawaii? :) Its such a shame I can't leave well enough alone and suck it up on POTS (eck). I'll keep you informed as to my progress (or lack there of). From my research, the problem with PRI's is that you generally pay a lot for the circuit - especially if you only need 8 channels or so. While you can go ahead and get a full PRI for not much more than a partial PRI, the cost of the taxes on the unused channels kills the budget when you look at a 2 year cost. I found a telcom broker in San Francisco that works with all the top providers, and while a lot of the competitors to the ILEC's had lower up-front prices, they got you by not including the costs of the taxes and had other fees too that killed the savings. This was especially true for less than 2 year commitments. Telco's hate BRI's because it takes more cable pairs and repeater hardware when you need more than one. In some cases they end up putting in a T1 / micro DLC. You also can't do DSL over an ISDN BRI line. BTW, a little birdie told me that Sangoma is working on a BRI card. Yeah! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk BRI in the USA
On Wed, Apr 12, 2006 at 09:10:09AM -1000, Mark Coccimiglio said: I guess what I need to find out first if there is anyone out there using Asterisk BRI in the USA? If so what hardware have they been able to use. I no longer want to hack around with analog circuits. BRI has the potential of PRI with only 2 B channels. A great idea for a small office such as my own. VoIP may be an option, but I would need a ITSP that would allow calls to transfer from my asterisk box to the remote phone set. My link to the internet is fast, but its pointless to route a call into the office just to stream it back out. More work more work more work. I'm in a similar situation. Being on the end of a long loop, POTS sucks - echo / static / crappy calling features. Paying around $2K-3K for BRI solution is a non-starter though. It needs to get down to the $200-400 / port level (more ports = cheaper per port) to be viable. Soho / Very small business (under 12 people) is definately a 1-2 port market which my guess would be the bulk of sales for BRI. It would be awesome to see a Sangoma BRI card. It's hard to say what the market would be since the US telco companies have really tried to kill BRI service. Considering what a full PRI costs, there is also a point where too many BRI ports no longer makes sense, but that number is probably 4-6 BRI's. I was in a situation where I really only wanted 4 BRI's, but had to look at a PRI instead which ended up wasting a lot of money in the long run. POTS was a non-option. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
On Sun, Mar 26, 2006 at 08:03:55PM -0600, Darrick Hartman said: Denis Galv?o - iSolve wrote: The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. WHAT! The Polycom phones that have speaker phone features (the 50x/60x) are great speaker phones. The 301 is not an speaker phone. It only has a listen-only hands free setup. In fact, the speaker phone is so good, most people can't tell that I'm on a speakerphone and are surprised when I tell them. I regularly use the phone both as a local conference phone and as a member in a conference with other people on speakerphones too. No issues at all, and great sound quality. The only conference phone I found that is better is one of the dedicated conference phones such as the 4000 or the old analog versions of it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 Endless Loop
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] said: I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers. Anybody have an idea as to how to get around this? I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). Since you are running Asterisk, it would make sense to use a Linux based FTP server. At least then you would have decent logging (turn on verbose logging) which you can post the output of. I would also suggest sniffing the FTP attempt with ethereal or tcpdump to get more info on it. In any case, you are going to have to get more details: When you say session, is it actually logging in correctly? Finding the files it is looking for? Or is it just a connection attempt? My guess is that it either is not logging in correctly or is not finding the files it wants, or it IS finding a file but doesn't like it. Possibly one or more of the files is corrupt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Dell blade servers
On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said: On Friday 06 Jan 2006 08:11, Richard Scobie wrote: Supermicro do not do Opteron (or Athlon64) systems. Supermicro DO do Opteron. Model numbers please? Searching through SuperMicro's web site shows ZERO AMD based models. ONLY Intel. They do have a few chassis that claim to support AMD based motherboards, but NO superservers or motherboards. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latest Source
Install Quotefix. Google is your friend. On Tue, Dec 20, 2005 at 10:56:13PM -0500, C F said: In M$outlook click on Tools Options select Preference then Email-Option then play around with on Replies and Forwards. Again you forgot to RTFM. On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote: I would if I knew how... Fraid I'm spending all my time on Asterisk, and not enough on Microsoft Outlook. No idea how to turn this on in Outlook. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tue 12/20/2005 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Latest Source Doug, Can you turn on indenting on replies? Your emails are hard to figure out who is saying what. Thanks, Steve They still support cvs? I was reading their patch docs (not that I want to make a patch), but it said to use: [EMAIL PROTECTED] ~]# svn checkout http://216.27.40.102/svn/asterisk/trunk asterisk and I get: svn: PROPFIND request failed on '/' svn: PROPFIND of '/': 200 OK (http://216.27.40.102) Had to use IP address... no DNS on test box... Doug -Original Message- From: Tony Hoyle [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Latest Source Douglas Garstang wrote: No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never used it... is there a Linux client for it? I've just done an update now and it works fine... Are you using the right settings from the website? (:pserver:[EMAIL PROTECTED]:/usr/cvsroot) Changing would kinda suck for me as I'd have to rely on tarballs from then on.. for various reasons the servers I build on only have cvs installed, and that's not likely to change in the future. Tony ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
Why oh why would you want to install *, which runs on Linux, on a machine made by a company that does NOT support Linux? Both IBM and HP do a pretty good job of supporting Linux. So do other Linux oriented companies like PenguinComputing.com Digium cards have historically been a little finicky in regards to which machines they work in, but Sangoma cards should work in virtually any modern machine that has the right type of slots (careful with some modern servers that ONLY have PCI Express slots.) Hopefully someone can comment about modern digium cards in regards to compatability. Have they gotten better? On Mon, Dec 19, 2005 at 08:44:38AM +0800, Hiu Yen Onn said: Then, how about Acer? Does it work well with asterisk? Simone Cittadini wrote: Matt Florell ha scritto: The best Dell for a production environment Asterisk server is no Dell at all. They make some great workstations, but I've had many problems with their servers(as have many others on this list) when trying to use them in production for Asterisk. Take a look at the Digium compatibility list: http://www.digium.com/index.php?menu=compatibility *I've installed a PowerEdge 2850 - Xeon 3.0GHz/2MB, 800FSB with two TE410P in it, the cards didn't worked out of the box, but they worked after a couple of hours googling around, and it is in production since 3 months, never gone down. * *(I'm not advocating dell, actually I don't even like dell as a society, only sharing my experience) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
On Thu, Dec 15, 2005 at 06:11:07AM -0600, Rich Adamson said: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? There might be a way for you to address your objective depending upon exactly what you're trying to do. The previous responses to your question _assume_ that each room in your case has a pbx extension (regardless of whether its a sip or analog phone). If their assumption is correct, then the responses are correct. However, if you want to use your existing analog phones and you group them together, several analog phones can share a single extension and those phones in the group can pick up and join the conversation whenever they want. Think in terms of using something like a Sipura sip adapter (or the equivalent from other vendors), and connecting all analog phones within your defined group to the rj11 analog jack of the adapter. One system I found that works well in a home environment is using a two-line, multi-handset cordless phone system. Run 2 analog ports to the base station, and this handles most home needs. Two users can make or receive calls, join existing calls, etc rather easily. The dial plan is set so that either line makes outgoing calls over a VoIP service, line 2, or whatever, so that the main incoming line is always available to receive calls. The home office has a Polycom 601 with it's own lines and dial plan logic, plus the fact that the polycom user is much more likely to know how to answer, transfer, park, etc. Wife proofing a * system is non-trivial and takes careful planning. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ip phone
On Fri, Nov 18, 2005 at 09:00:20AM -0400, Doug Meredith said: stevanus [EMAIL PROTECTED] wrote: Maybe grandstream budgetone 100 series will fulfill your requirement. It's very good for such a cheap sub-50 phone. We have two of these and they are the VoIP equivalent of a $10 K-Mart phone. I won't even use them in my house, much less the office. Yep - I have one in my junk box. Maybe the SPA-841 would be a better choice for a few dollars more (haven't played with one personally, but everything I've heard says that they are much better than the GS BT's.) I'm not a fan of analog phones. Except for lobby, kitchen, or conference room phones, anything less than 2 line appearances is a PITA in the business world. A single line phone (even with *) makes it difficult to (for example) put someone on hold, call someone else to ask a question, and then return to the primary call. This means that each analog phone would need 2 ports off a channel bank. New pricing for a channel bank (ADIT 600) runs about $3300 for 48 FXS ports (I don't know why people keep quoting ebay prices... Let's be real here. If you can find them new for less, please let us all know where.) 2 48 port boxes with a 4 port Digium echo canceller quad T1 card will run you around $88 per port PLUS the cost of the phone - and a good analog phone is going to be a minimum of $50 for a single line version, $89 for a two line. This puts us at $138 for a single line and a whopping $265 for a 2 line phone by going analog. When you can get a Polycom 501 for $199 qty 1, it obviously doesn't make sense to use 2 line analog phones at all. With the 301 at $130 (froogle shows as low as $106), it doesn't seem to make much sense to use analog phones at all. There are Many sub $100 IP phones that are pretty good as well, which tosses out the reason to even maintain existing analog phones. The 301 has an ethernet switch, so chances are you don't have to rewire at all (so that argument is moot in most cases.) There are a few cases where analog phones may still make sense - door phones, conference phones (if you have an existing good polycom), etc. All in all, going analog seems like a pretty silly thing to do when you look even just a couple years down the road. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [OTAnn] Feedback
This is blatant spam. Looking at: http://roomity.com/advertising.jsp it looks like they have spammed at least 60,000 other mailing lists too. WTF would I want to use their crappy video and flash ad spewing crappy web interface that requires me to be online all the time over my awesome ad-free threaded client? If you run a mailing list or email server, it's time to firewall their ass. On Tue, Nov 08, 2005 at 10:16:18AM -0500, Steven said: I use a newsreader pointed at gmane.org. It is agregated and only uses my internet connection when I tell it to. shenanigans [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I was interested in getting feedback from current mail group users. We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband. Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [OTAnn] Feedback
I woudn't see why not. I have not seen a mail server / firewall that can't ban a netblock. Turns out this a-hole has now spammed several other lists I'm on. If we get enough people complaining to hurricane electric ([EMAIL PROTECTED]) they won't last long. On Tue, Nov 08, 2005 at 02:21:58PM -0500, Kanuri, Seshu (Company IT) said: I totally agree. But doe the Asterisk list servers have any such feature to block the spam and delete the spamming users? I don't think so. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Tuesday, November 08, 2005 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [OTAnn] Feedback This is blatant spam. Looking at: http://roomity.com/advertising.jsp it looks like they have spammed at least 60,000 other mailing lists too. WTF would I want to use their crappy video and flash ad spewing crappy web interface that requires me to be online all the time over my awesome ad-free threaded client? If you run a mailing list or email server, it's time to firewall their ass. On Tue, Nov 08, 2005 at 10:16:18AM -0500, Steven said: I use a newsreader pointed at gmane.org. It is agregated and only uses my internet connection when I tell it to. shenanigans [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I was interested in getting feedback from current mail group users. We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband. Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double DTMF with tdm card
Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM0xB vs. SIP for FXO
I've had issues with the FXO port on the spa3000 - banking apps could not hear the DTMF. I've also had problems with phones hooked up to the TDM FXS ports where banking apps hear DOUBLE dtmf digits. The only mix that seems to work for me is SIP phones / or analog phones hooked up to ATA's and TDM / x100P connections to the POTS line. Frankly, this situation sucks. I've got half a TDM card that is unusable, and half an SPA-3000 that is unusable. The SPA was really attractive since it would have allowed for automatic powerfailure mode. I dumped my X100P's since the telco tech showed me that it was generating almost a direct short across the line according to his meter. The TDM card doesn't have that problem. On Wed, Nov 02, 2005 at 11:45:20AM -0500, Rusty Dekema said: Hi, I am planning to connect my Asterisk PBX to one or two POTS lines, and am wondering if it is better to use a TDM card for this, or one or two SIP devices with FXO ports on them (such as an SPA-3000, Grandstream 488). I am interested in voice quality and reliability of operation and am wondering if one of these options is better than the other in this regard. Thanks, Rusty ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double DTMF with tdm card
Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double DTMF with tdm card
Frankly, I think this may be happening to me too. It's still a zap to zap channel problem. On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said: My problem is slightly different as there is 2 T1 Ports involved - With a T1 test set I can clearly hear two tones sent quickly with each outside caller press. I assume one of the tones is the actual audio passing thru the connection and the other generated by the T1 card itself.If I make the same test with a TDM400 as input connection and the TE410P port as output connection, there is no double dialing. Same results if an inside extension is used as input connection. It only happens if it's a T1 to T1 Bridge... If it is a regenerated tone from the TE410, it seems there should be some option to stop listening for tone touch after connection has been established? Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Eric ManxPower Wieling [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:50 AM Subject: Re: [Asterisk-Users] Double DTMF with tdm card Note this is on external calls to external applications Not Asterisk DTMF detection. It's as though DTMF is distorted when going through a TDM fxs port, or that it's being caught (too late) and then retransmitted. Does * intercept outgoing dtmf? I haven't found good docs that tell exactly what relaxdtmf does. On Thu, Nov 03, 2005 at 08:01:03AM -0600, Eric ManxPower Wieling said: Did you try relaxdtmf=no Walt Reed wrote: Nope - I saw your posts on it though. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double DTMF with tdm card
I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)
On Tue, Oct 18, 2005 at 09:10:38AM +0200, Tzafrir Cohen said: On Mon, Oct 17, 2005 at 07:01:17PM -0400, Walt Reed wrote: I was unable to get a clean compile of the kernel or * with gcc 4. You can ask about this in Debian lists. I don't have unstable so I can't test for myself, but the current unstable kernel surely builds for them. Sure, but it depends on which compiler they use... They may not use 4 on the kernel. In fact, I'm pretty sure they don't. The version of the kernel I was compiling is the version WITH debian patches (latest.) And it surely does not compile with gcc 4. As for Asterisk 1.2: It should hit experimental any day now. There are also unofficial debs at http://rapid.dotsrc.org/experimental/ . If those don't build with gcc 4 then this should be reported. Most gcc 4 incompatibility bugs I saw were fixed pretty fast. The incompatabilities are going to be kernel compiled with one version of GCC, zaptel module compiled with another. That, even if it appears to work, is not a good idea. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] huge problem compiling * on gcc4.x (SUSE 10.0)
On Sun, Oct 16, 2005 at 09:21:09PM +0200, [Ludwig IT-Services - GMAIL ] - Michael Ludwig said: I'm very new to this list and to asterisk and stuff at all. To build my asterisk server I installed a new machine running the new SUSE Linux 10.0 (retail version on DVD). I need asterisk (tried 1.0.9), bristuff (off junghanns.net, -0.2.0-RC8o) and the florz-patch because I have two HFC-S-ISDN cards in that machine. Now when it comes to compiling I get a huge bunch of warnings and stuff, zaptel 1.0.9.2 fails to compile and asterisk 1.0.9 also fails to compile. SUSE 10.0 uses gcc 4.0.2 and as I asked in some other mailing list and forums, that is the reason why * stuff fails to compile. Is there any stable asterisk version available which does compile fine on a gcc4.x ? If not, will the * source be changed to finely compile on gcc 4.x? If yes, when will that be? (I need the * stuff now). If not, why not? What's on with the 1.2.0-beta stuff out there on the asterisk.org webpages? Does that one compile on gcc4.x ? I've been running a * (cvs HEAD) instance on Debian unstable, which has upgraded to gcc 4. Gcc 4 still has problems compiling the kernel (as of 2.6.12) on debian, and you want to use the same version of the compiler on the zaptel modules that you do on the kernel. I was unable to get a clean compile of the kernel or * with gcc 4. The good part is that I have gcc 3.3 on the system too, so it's an easy symlink change. It may be that gcc 3.x is available on suse either by default or an additional package. Considering how new gcc 4 is, and how many major changes there were with it, I personally would wait another 6 months to a year before using it in production. I've also read stories that gcc 4 produces slower code than gcc 3. I'm sure others will have some insight as well :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Sangoma AA Series?
On Thu, Oct 13, 2005 at 12:25:51AM +1000, Mark Lipscombe said: This is at http://www.telephonyware.com/sangoma In the mean time, here is some more information so this thread hasn't been a waste of time. The new cards will be available soon, and will also have an option for an addon 16 port hardware echo canceller with a 128ms echo tail -- this will be available in early December. The analog boards will consist of three components, a shark board, which is the base PCI board, a daughterboard, and FXO/FXS modules (with two FXO, or two FXS interfaces per module). The first board you have in your system will have the base board, the daughterboard, and one or two FXO or FXS modules. When you want to add more than 4 ports, you add another daughterboard, and one or two more FXO/FXS modules. These basically sit over a PCI slot, and screw into the back of the chassis, but do not actually sit in the PCI slot itself. Everything is then connected via an external backplane, in much the same way a series of SCSI drives are plugged into a daisy chain. I don't quite understand why a single-card solution is being avoided by Digium / Sangoma. This solution is interesting, as it looks like it is designed to fit small form PCI (for 2U servers) but riser cards in servers like the HP DL380 and others make this less of an issue. I can get 3 full length / height cards in a DL380. If they offered a single card 12 / 16 port version (using 4 port modules,) they should be able to keep the per-port cost down, and the slot count down without the goofy backplane. It also looks like each daughtercard needs it's own drive-power cable (from the picture.) The 12-16 port market is really underserved - too small for T1 / channelbank, too large for solutions like this. I like alternative vendors, and Sangoma's reputation is very good, but this product looks like it was designed as a solution looking for a problem. If I want 16 ports, I need 4 slots worth of room. Using a 2U server (like this card looks to be designed for) I could only get 12 ports max anyway (as most 2U servers are not going to be loaded with slots.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which asterisk-friendly cards are fax-capable?
Hi All, - I have Digium cards and given that the archives point out the Digium cards drop packets does anyone know what hardware would not do this? (i.e. Allow me to send outbound faxes) - If there is still an issue with the wctdm driver, does anyone know which asterisk/spandsp combo would work for sending outbound faxes? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax (app_txfax) sending issue
Hi All,
With spandsp.0.0.2 pre20 installed I can't seem to send faxes with
tx_fax over a Zap channel (POTS). rx_fax works just fine so no issues
with libtiff and (presumably) libxml2.
Basically I get 'slow carrier up' and 'slow carrier down' together with
accompanying beeping noises until tx_fax times out and hangs up. This
could quite possibly be a PEBKAC or n00b problem since I'm relatively
new to Asterisk.
- snippet: extensions.conf -
exten = _8NXX,1,Set(FAXFILE=/var/spool/asterisk/fax/sendfax.tif)
exten = _8NXX,2,Set(LOCALHEADERINFO=Company name and department)
exten = _8NXX,3,Set(LOCALSTATIONID=Company name)
exten = _8NXX,4,TXFAX(${FAXFILE}|caller)
- // extensions.conf -
- snippet: * console -
*CLI -- Executing Set(SIP/rec1pub1-af67,
FAXFILE=/var/spool/asterisk/fax/sendfax.tif) in new stack
-- Executing Set(SIP/rec1pub1-af67, LOCALHEADERINFO=Company name
and department) in new stack
-- Executing Set(SIP/rec1pub1-af67, LOCALSTATIONID=Company name)
in new stack
-- Executing TxFAX(SIP/rec1pub1-af67,
/var/spool/asterisk/fax/sendfax.tif|caller) in new stack
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
...
- // * console -
Could anyone tell me what I might be missing or what I have to look for?
Thanks in advance.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
Or shitcan the onboard raid and get a real hardware raid controller like a 3ware card (if you are stuck on IDE / SATA.) Reduces complexity. On Fri, Sep 09, 2005 at 09:23:44AM +0300, Tzafrir Cohen said: On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. Also: in such a settings you can get comperable performance by using Linux's built-in software raid. And for that you won't depend on non-standard drivers from the vendor for that. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CD copy
And this has to do with Asterisk exactly how?? This is not a MS Windows support group. On Fri, Aug 26, 2005 at 06:42:33AM -0700, Ellafi Fituri said: Hi, I have 2 CDs that would like to make a backup of , I am having a hard time doing. I have tried NERO ver.6 but it does not work it always report unrecoverable sector. Does anyone knows of a copy tools to use to copy the CD Any help will be very nice and appreciated. Thank you all... Ellafi - Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
On Tue, Jun 28, 2005 at 10:59:31AM -0400, steve szmidt said: On Monday 27 June 2005 20:04, Robert Webb wrote: I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. In a Google search box: site:lists.digium.com What you are searching for The problem many newbies faces is TOO MUCH information. Not being able to see the trees because of the forest basically. It does not matter either if it has been discussed until someone went crazy or died. The reason it keeps coming up is because it has not been solved. The problem is NOT that the archives are not searchable or indexed. The problem is that we are dealing with a very complex subject / application. Telephony is Very different from general computing. The terminology, technology, typical problem set is unique to the industry. Now we add all the typical computer and network issues on top such as IRQ's, QoS, firewalls, NAT, etc. Newbies don't have a chance unless they are willing to spend the time it takes to learn about the technology they are trying to implement. Maybe some day this will change as hardware gets better, easier to configure, and the software matures. People that want something that just works without tinkering NOW should either be NOT using VoIP technology, or should be purchasing a complete solution (or consulting services) from one of the many vendors available. * and VoIP are still very young, about where Apache was as a web server (and the internet in general) back in 1996 - pretty stable, but still immature. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org unreliable lately?
I doubt it's the software itself (I run Tiki too... It's just PHP.) It's purely a matter of scaling. What part is causing the load? The PHP apache processes? The DB server? Both? What performance tuning has been done? Is it a custom apache compiled for this app or is it a generic distro version with all the extra modules? What about the disk? Is it fast enough (slow disk can cause high load average numbers as you spend all your time in I/O Wait.) Is there enough RAM? What else is the machine used for? Whatever is causing load problems can be analyzed and solved. May need hardware thrown at it. May not. In any case, scaling tiki up is easy - all it takes is time and money. You can throw multiple front-end and back-end servers at it to handle any reasonable load (we are not talking Yahoo levels here...) Converting the content (as is the case with many Wiki's) to another wiki system can be quite painful. Each uses their own wiki notation and has their own list of features. Converting is usually a last resort. On Wed, Jun 22, 2005 at 01:29:35PM -0500, Jay Milk said: As I understand it, the wiki software behind voip-info is not able to keep up with the load. There may be better (-performing) alternatives such as MediaWiki, but the question would be that of conversion... And whether the owner even wants to convert. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info.org unreliable lately?
On Wed, Jun 22, 2005 at 01:46:39PM -0700, Frank Mayhar said: On Wed, 2005-06-22 at 16:05 -0400, Walt Reed wrote: (slow disk can cause high load average numbers as you spend all your time in I/O Wait.) Um, no. At least in traditional Unix (meaning System V and the BSDs), the load average is the average length of the run queue. By definition, if a process is asleep waiting for I/O to complete (as is the case for disk), it is _not_ on the run queue, and so doesn't contribute to load average. On linux (at least RHEL3) it sure as heck does. I see it VERY frequently on my RHEL3 boxes - especially when dealing with huge postgres DB's (300G). For one application, moving off local disk onto a SAN sped up the I/O and my load averages went from 40 to .2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Coding a telemarketing call blocker
On Thu, Jun 16, 2005 at 10:45:04AM -0600, Tore Hansen said: I am interested in creating a telemarketing call blocker in my Asterisk dial plan. I am not much of a programmer, and I am wondering if external AGI code would be required to implement this. The logic that I would like to have in place is this: 1. If the incoming call carries proper name and number caller ID, then ring default extension. 2. If the incoming call carries no caller ID information, then send call to recorded message, followed by voice mail. 3. If the incoming call carries number only caller ID (no name info), then check the area code the call is from. If it is my local area code, then ring default extension, but if it is from a different area code, then send call to recorded message, followed by voice mail. See the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoIf Example 3 has some logic that could easily be extended to do exactly what you want. With modern versions of * (CVS HEAD for example) the dial plan can be simplified a bit. I wrote that example pre 1.0 days... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [ADMIN]: subscription failure
Did you go to the web page that is listed at the bottom of every message? Look at the bottom of that web page for the address of the list admin. By the way, that admin address is pretty much standard for ALL mailing lists. On Wed, Jun 08, 2005 at 07:37:35PM -0700, David Koski said: Would an admin please contact me off list? I tried to subscribe from another address and it failed--I got no email to confirm the subscription. I would rather use the other address and need to know if there is a problem with my mail server. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Station Lines
On Wed, Jun 08, 2005 at 08:38:27AM -0400, Sean Cook said: The feature that he really wants is to be able to pick up any line and have all the stations show up on his phone. Is this possible in asterisk? If so can someone point me in the right direction? That describes a key system. Asterisk is a PBX. Trying to make Asterisk function like a key system (while possible) is difficult, and will result in much frustration. Instead, you are best off showing this person how to use a PBX properly. It may take a little getting used to, but it's best in the long run because you won't be supporting a goofy system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
See http://www.windsun.com/Batteries/Battery_FAQ.htm Even deep cycle marine batteries can be crappy, with many actually being a combination of starting / deepcycle. The primary difference is the plates. Good deep cycles have thick lead plates. Industrial batteries should last longer. Car batteries typically have sponge style plates which offer high CCA but low life in deep cycle applications. UPS applications are very hard on batteries unless your power is REALLY bad, and goes out all the time. Lead-Acid batteries need to be used. On Tue, May 31, 2005 at 04:17:41PM -0400, Nick said: Use deep cycle marine batteries or similar. Car batteries arn't really designed for long low power, they're designed for CCA, high output short burst. Nick On Tue, May 31, 2005 at 01:22:09PM +0400, Jean-Michel Hiver wrote: Another thing to consider regarding the ups is the runtime, depending=20 on the hours and minutes you want the ups to supply power to your=20 asterisk box, you may need to add more batteries to the ups. Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get=20 monstruous UPS capacity... the only trouble with it is that re-charging=20 the batteries may take some time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play gsm files in windows
On Mon, May 23, 2005 at 03:34:44PM +0100, Brett, Gary said: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. See the Wiki: http://www.voip-info.org/wiki-Asterisk+sound+files ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card Line-In as MOH source
On Thu, May 12, 2005 at 02:53:02PM -0700, Chris Coulthurst said: Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. You can probably use sox as a filter / source, but it will probably take code. Why would you want this? Just record the tape / CD to a file... If you are thinking of piping radio, that probably violates copyright and you would also need to pay ASCAP fees... BTW, posting your question twice half an hour apart is not good etiquette. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting variable for a context for all extensions?
On Mon, May 09, 2005 at 03:28:22PM +0200, Mark Wormgoor said: Hi, Is it possible to set a variable for a context for all extensions? I haven't been able to find it. Try looking up the application SetVar: demo*CLI show application SetVar demo*CLI -= Info about application 'SetVar' =- [Synopsis]: Set variable to value [Description]: Setvar(#n=value): Sets channel specific variable n to value But how do I link SetVar() to all extensions in a config? If I use exten = _.,1,SetVar() it will never continue on exten = 1234,1,Dial or exten = 1234,2,Dial. Use dbput/dbget. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring
On Sun, May 08, 2005 at 12:15:09AM -0500, Anton Krall said: How do you configure asterisk to recognize distingtive ringing using x100p cards? Can this be done and how? Check the example in the zapata.conf file, and the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
On Fri, May 06, 2005 at 04:24:32AM -0500, Eric Wieling aka ManxPower said: Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. Not quite. A pure life-line FXO that is not voip accessable is useless to *. Usually this means that an extension on the FXS port uses the PSTN on the FXO during powerfailure / 911 calls. Some ATA's have this kind of port. The SPA-3000's FXO CAN pass through in life-line mode automatically for power faliures and if it is configured to do so via the dial-plan. The dial plan on the 3000 allows lots of flexibility here. From a VoIP standpoint, the FXS and FXO ports can be configured to be totally separate devices, where if you want to make a call via the PSTN, the call is looped through *. Pass through can also be used in terms of how the FXO interfaces with *. The standard config of the SPA-3000 for example answers the call and THEN forwards to * - acting more like a full gateway than a dumb FXO. It can also be configured (kludged) to pass through call info to * BEFORE the call is answered (which is frequently more desirable in many situations.) Hope this helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 x TDM400P in one PC ??
On Fri, May 06, 2005 at 10:25:24PM +1000, Joerg Wleklik said: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Does anybody have experiences with plugging 3 TDM400P cards in one PC?? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. That's what the channel bank does. The other reason you want to use a T1 card over multiple TDM400 cards is that the 3 TDM cards will generate 3 times the number of interrups, and likely have interrupt sharing problems. Good channel banks also are going to be much less prone to have echo problems. You also will have room for expansion. BTW, if you went with a new T1/PRI to the telco, you can probably have your old numbers forwareded / migrated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 x TDM400P in one PC ??
Reformatted top-posting... On Fri, May 06, 2005 at 08:30:52AM -0600, Andres Paglayan said: Walt Reed wrote: On Fri, May 06, 2005 at 10:25:24PM +1000, Joerg Wleklik said: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Does anybody have experiences with plugging 3 TDM400P cards in one PC?? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. That's what the channel bank does. The other reason you want to use a T1 card over multiple TDM400 cards is that the 3 TDM cards will generate 3 times the number of interrups, and likely have interrupt sharing problems. Good channel banks also are going to be much less prone to have echo problems. You also will have room for expansion. BTW, if you went with a new T1/PRI to the telco, you can probably have your old numbers forwareded / migrated. Why the channel bank if he will be routing extensions to ip phones? The T-1 card should suffice if he isn't serving analog extensions. Thats $600 (t1) instead of ~ $1000.(3 x 3?? tdm400p) line cost wise 12 channels on a t1 should be cheaper than 8 pots. Because he STILL needs analog lines for the legacy PBX (read above), and he does NOT have a T1 now - just POTS PSTN lines. That's 12 analog ports needed. Now if he were to convert to a T1 (or E1 / PRI) for his PSTN connection (as I mentioned,) then he could get a couple cheap Sipura's for the legacy PBX and forget the channel bank. Depends on what he can get for T1/E1/PRI pricing (8 channels is usually not very cost effective.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Review Outgoing VM Messages
On Fri, May 06, 2005 at 02:02:12PM -0400, Christopher Jacob said: Hey All, I had a user ask how to go in and listen to her current outgoing messages. I must confess, I can't figure out how to. Any ideas? I don't believe there is a way. It would NOT be hard to add a review feature. I believe the sound files already have the verbage to support this IIRC. IMHO, the existing VM app is pretty weak and has a number of deficiencies. Someone else here re-wrote the entire voicemail system as an AGI with advanced features. I think I read about it earlier this week on the list... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
On Fri, May 06, 2005 at 11:49:42AM -0700, Rusty Shackleford said: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Friday, May 06, 2005 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FXO ATA? Why not go with Multitech? They are expensive, but great units. Because they are ridiculously expensive. It is true that Multitech's VOIP gear is very good stuff. I've used it and it just works. But apparently, their marketing people haven't been paying attention to the market and they are still using pricing that reflects the market 5 years ago. Multi-tech has always been this way across their entire product line. They sell enough units to stay in business, but are priced in a way that ensures that they will never be a market leader (in terms of unit sales.) It's too bad, because technically they are awesome. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Review Outgoing VM Messages
From looking at the source, I don't see anything that allows a review without changing the greeting first. In the function vm_options(), it calls play_record_review() when you choose a greeting. play_record_review explicity sets cmd=3 with a comment Want to start by recording. Command 3 starts recording immediatly after playing the intro and beep. There is no option to review first. I think some of this logic is due to the fact that the same code is used to record callers messages as well as greetings, and the code is geared towards callers more than greeting setting. It shouldn't take a whole lot of code to fix this, but it may be easier to copy the play_record_review() function and create a specific version for greetings than to fix this one to handle both cases nicely. It's a pretty short and simple function. On Fri, May 06, 2005 at 04:45:35PM -0500, Anton Krall said: I just tried that :) and there is only the modify options... |-Original Message- |[mailto:[EMAIL PROTECTED] On Behalf Of |Colin Anderson | |1. Log into Comedian Mail |2. Press 0 for System Options |3. In there you can hear/modify Busy, Unavaliable etc messages | |I know what you mean, it isn't obvious from the prompts how to |do it, first time I did it I had to screw around for a while. | |hth | |-Original Message- |From: Christopher Jacob [mailto:[EMAIL PROTECTED] | |Hey All, | |I had a user ask how to go in and listen to her current |outgoing messages. I must confess, I can't figure out how to. |Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux
On Fri, May 06, 2005 at 07:43:15PM -0400, Jim Archer said: Hi All... I have an Asterisk 0.7x server running and have forever now. I would like to upgrade it to 1.0 (or whatever the current version is). It's running on Linux. I have been told there is now a Debian package for Asterisk on Sarge! I was looking at the Asterisk web site and I noticed that the Wildcard X100P cards are deprecated. I am using two of these cards to interface to POTS lines. If I upgrade to Asterisk 1.x, will I still be able to use these cards? Are there better cards I should look at that will improve quality or offer more features? Yes. The old X100P cards still work fine (in the US, in most cases) with both 1.x and cvs HEAD (the dev branch.) That said, I'm migrating a similar setup to one X100P and one SPA3000 to cut the number of interrupts in half and free up a slot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux
On Fri, May 06, 2005 at 08:02:43PM -0400, Jim Archer said: Thanks Walt, that's great! You just remindedme about something, although I don't know why. When I first set this up, I wanted Asterisk to detect distinctive ring patterns and only answer a particular pattern, so that I could share a fax line. At the time, it was not possible. Has this changed? Will new hardware do this? Yes and no. I had dring setup, but the problem was that I had dring on both line 1 and line 2, and the code had no way to specify different contexts on a per channel basis. This has not changed AFAIK. If you only have dring on one line, and the fax number is a dring number (not the primary number), then I could see it working. Otherwise, probably not. Dring is really only useful on single ZAP FXO boxes. The SPA-3000 does not do dring detection on the FXO port IIRC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
On Thu, May 05, 2005 at 12:33:50PM -0600, Joseph said: The difference is that SPA-3000 answer the phone and rings asterisk (the phone at this moment has been answered the ringing party is incurring the charges before asterisk answered the phone), the AG-168 is ringing the asterisk directly, so I think the pass through port is a benefit in this case for asterisk users. See here on how to pass through with the Sipura: http://voxilla.com/forum-viewtopic-t-1335-sid-c3365f7a694970ed5b7fa0fce2618636.html Yes, I've tested it and it works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avaya 4610SW IP phone?
From what I've read, this is a H.323 phone only. Only the 4602 has SIP images. Has anyone gotten a 4610 H.323 working with *? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this
On Fri, Apr 22, 2005 at 02:40:10AM -0500, Paul said: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I can make a call from a POTS phone hooked up to a Cisco ATA 186, out one X100P to the PSTN, back in a second X100P, to a phone hooked up to the second port on the ATA186 with no noise, and no echo, and a pretty small delay (which you can hear with one handset in each ear.) I have disabled most of the on-board I/O such as parallel, serial, and extra USB controllers, and the X100's are on int 5 and 7, not shared with anything. Interrupts 10 and 11 have a bunch of stuff shared and are used by USB controllers, ethernet ports (one on each IRQ) video card, SCSI controller, and one unknown device (some special nVidia device.) This machine is also used as a firewall / gateway / email server but does NOT run X (which I hear can cause problems on some machines.) I've been running this configuration for about 9 months with virtually no problems in a SOHO environment including weekly 3-hour long conference calls. I realize this doesn't help you much, but it IS possible for the configuration to work. I have been thinking about getting a Sipura 3000 to add another FXS port and remove one X100P which would also cut down on the number of interrupts, leaving me one X100P for timming (so I don't need ztdummy.) MAYBE this would help you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and ease of use would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. I have worked for over 10 years in the software development industry and Then you entered the industry far too late to know the real history of computing, have read too many MS revisionist history books, or were hiding under a rock. For example, The Amiga for example had a wonderful OS, great multi-tasking, awesome windowing interface etc. over 10 years before MS (some would argue longer.) Comodore didn't have a chance against the mighty combo of IBM, MS, Compaq. and other x86 hardware and software vendors in the business world (the Amiga was originally designed as a game machine and could never escape the stigma AND had the same bone-headed single hardware source issue that Apple has. Poor management / marketing also contributed to the companies death.) (Speaking of Apple, it boggles the mind that it took them over 15 years to add multi-tasking to their product line - and yes, I am dismissing their prior failed unix attempt.) MS has no effective competition due to their illegal business practices, killing off alternatives (BeOS is a recent example) by pressuring large ISV's to only write for the Windows OS, restrictive contracts with hardware vendors, and other sleezy tactics. They effectivly killed Java on the desktop. They continue with a powerful FUD campaign against Linux, Apple, Firefox, etc. I could go on, and on, and on. In my opinion, MS has held the world of computing back about 15 years (unless you think that having the worst security model / track record in computing history, and proprietary interfaces and file formats with no publicly available documentation is a good thing.) Unfortunately the reality of business means that we have to deal with this horrible corporation and their aweful software. MS and their single platform (for servers and desktop anyway) means that we are still saddled with the horrible x86 architecture, the interrupt structure, bus, bios, etc. (essentially most everything about a PC.) By the way, that architecture is why it's so hard to make reliable hardware, why we need an external card to get a reliable timer device, etc. Before you spout off about how great MS has been to the industry, maybe you should learn a little about that industry and it's history first, M-kay? /rantmode ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
On Wed, Apr 20, 2005 at 09:01:56AM -0700, trixter http://www.0xdecafbad.com said: On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote: On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and ease of use would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. and hoiw many operating systems were so popular during the 80s and early 90s? What operating system shipped on almost every computer during that period? BTW, in the 80's, it wasn't windows - it was DOS (I know, well before your time.) Again, nobody could really compete with the IBM / MS / compaq x86 platform dominance, so the ONLY real choice on that platform was Dos, although there were a few specialty OS's and extensions (OS/2, QNX, Desqview/X, etc.) I realize you wouldn't know about them, comming into the game rather late. It wasn't until Windows 3.1 in the early 90's that there was a relativly stable (if you could call it that) windowing system from MS (despite that other companies had been doing it for many years.) Bundling and restrictive contracts made it impossible to compete. Furthermore, (if you knew your history) MS had been doing funny things with DOS / and windows to make it difficult for other windowing systems and DOS clones to work with MS-DOS / Windows, further cementing their market dominance. I dont think I lack understanding of the industry I think that I remember clearly that windows was shipped on that, I think that whether or not it resulted in an anti-trust conviction microsoft did make it easier for people to use computers and thus more sold. Again, your lack of experience with and knowledge of other OS's shows otherwise. I am sorry that you are so bigioted to think that other operating systems dominated the market during that period, and cant accept that windows was the #1 operating system by a clear margin in terms of installed systems. Did I say they dominated? No. Please work on your reading comprehention. There was competition on the OS front, but it's hard to knock out the market leader, and impossible when they won't play fairly (legally.) I have worked for over 10 years in the software development industry and Then you entered the industry far too late to know the real history of computing, have read too many MS revisionist history books, or were hiding under a rock. I started using computers in 1976. I dont think I entered too late. As for reading MS revisionist history books, no but I think that you have been readiung too many anti-MS revisionist history books. The popularity of a personal computer in the home was not made with cp/m it was not made with coherent (a unix for the pc before linux was around). It was not made by os/2, it was not made by any mac. Computers did not fully become so incredibly popular until windows. look at any historical sales reports and see when the numbers started increasing dramatically. Again, bundling, restrictive contracts, buying and killing your competition, sueing your competition, not working with standardsm etc. These are the things that created the dominance. You can't possible comprehend reality until you are willing to accept these facts. BTW, if you really started using computers in 76, in what capcity? Playing Pong? Recall all the software shops that sold software, why was it that at least 90% was for windows and the remaining 10% for all other operating systems for a great many years? Why did all the computer shows that were oh so popular during that period sell mostly for the wintel platform? That was not always true. If you REALLY have been professionally using computers since 76 (or even 1990) you would realize that this was not true until the early 90's. For example, The Amiga for example had a wonderful OS, great multi-tasking, awesome windowing interface etc. over 10 years before MS but it never sold as well. You fail to understand that its sales that drove the cost down. os/2 was better than windows at multitasking too, but again it didnt sell so well. Granted there was evilness by microsoft that resulted in antitrust convictions over some of that but you just proved how clueless you are. How many times do I have to say it? Bundling, restrictive contracts, unfair / illegal business
Re: [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?
On Sun, Apr 17, 2005 at 01:50:56PM -0700, snacktime said: On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote: I have been trying a did company for a few days. I find the service decent, but sound quality only moderate. Rather than spending 35 or so for monthly with did, I am considering an isdn bri at this location. How much more stable and reliable is bri or pri versus a voip did service? I like the concept of a bri more, but I do not get cid generation. Would anyone suggest bri over voip where available? I must say, I prefer higher voice quality. If anyone finds bri to be worth it (at about 54/month plus usage) please let me know what you think. I'm kind of asking the same questions myself right now. I think it depends a lot on what you are planning on using voip for. I also think that you are going to see reliability go up and up over the next year or two, so you have to take that into account also as you plan your infrastructure. I think new installations should at least be voip capable. No matter what the usage is, BRI / PRI will be more reliable. VoIP to a generic providor will never be as reliable as a dedicated connection to your telco carrier of choice. Now whether you can live with the level of reliability is another story :-) The big problem with with VoIP is lack of QoS beyond your local network. Probably the best situation is to get your VoIP from your local ISP where QoS can be implemented end to end. Other current VoIP issues include spotty Fax support and flakey SIP / IAX support - these should be resolved in time, but they are a big problem now (as the volume of emails on this list related to providor problems shows.) As for QoS support on ther internet in general, well, I wouldn't hold my breath, and that is what is really needed to increase reliability / sound quality. Right now I would not rely on voip 100% for something business critical. Personally I'm looking at using voip but having adequate pstn access as a backup, with the incoming DID numbers being able to automatically route to the pstn in case of failure.I know I can do this if my numbers are 800 numbers, but I've still not found a way to do this with local number DID's, although I'm still looking. Reliability on incoming lines is a lot more difficult to deal with then outgoing. As long as you * server has connectivity, you could have 4-5 different providers in your dialplan and have it cascade down through them on failure. Wish it was that easy with DID's. True, if the providor is totally down you can fail over, but if the providor is up but not working well, you will have sound quality problems, dropped calls, etc. and there isn't a good way of handling this at the moment (could probably handle this via some new * code to score a providor during a call and drop them from the list if there are too many dropped packets, etc.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 nor *8# works for me!
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said: Eric Wieling wrote: I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk What am I going to look for, e.g., in a manual for snom 190 and a Budgetone ??? See the Wiki: http://www.voip-info.org/wiki-Asterisk+config+features.conf I had the same problem with Cisco ATA's screwing with the *, so I changed mine to a normal number and everything works great. I never did figure out how to make the cisco pass the *8 properly. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Are there online forums instead of this
First, trim your posts. Why include extra copies of the footer? Does it help this discussion? On Fri, Apr 01, 2005 at 02:17:52AM -0500, Tim Bass said: I'm saying that as a long as long as Digium supports this dinosaur technology in support of their community that is exactly what the community will have, and nothing better, because this is the Digium supported community. The term better depends on your technical expertise and point of view. I know how to use my email client. The interface I have is better than any web forum software on the planet, and don't get mouse finger strain using it. Of course if you insist on using a brain-dead mail client (outlook comes to mind) you may find it frustrating. That's your fault - not protocol's. It is really obvious to an unemotional objective user who has reviewed the archives, the search function, Google works fine. Knowing how to use it is important though. If you won't learn how to use the tools, you won't be able to use them effectivly. and has observed the disorganized, helter-skelter, all over the map discussions Again, use a proper threaded mail client and topics are simple to browse. (ok, I guess, if you have lots of free time on your hands), poor text formatting messages (i.e. no way to indent code, code fragments, highlight, etc.) - Tab key must be broken on your computer??? Maybe your editor sucks? That's why messages look bad. Frankly, I don't want to spend all my time formatting a message. Formatting is eye-candy and has little real value. this helter-skelter community has a solid a one-hour post-to-message lag time for recent subscribers and traffic-volume that is not possible to moderate to enforce simple social rules and professional conduct. Those are hardware / bandwidth / list-maintainer problems. Not the protocol's. Performance is an easy fix. A web interface would have MUCH MUCH higher CPU / bandwidth needs. The software can also be configured to reject HTML messages, attachments, and any message containing multiple copies of the footer (which it should). A moderator can ban distruptive users as well. For example, vBulletin's (www.vbulletin.com/forum) entire business ecosystem is supported by very a very large community of very talented users and developers. Some of the top developers also support parallel ecosystems such as www.vbulletin.org/forum where customization is distinct from core services and basic user support. I find the sofware highly annoying - only using 1/4 my browser window width being the least annoying issue. The thread view only holds 7 messages before you have to scroll and is not proportional to the browser height. I could probably go on for pages on the annoying characteristics of that software, but the bottom line is that you are FORCED to use that one interface. With email, you can choose any interface you want, maintain your own personal archive, etc. These people are very top technical people (not some lamers who can't use email as some recent foolish posters have demanded) and they certainly could not support such a complex and sophisticated user community if they used an antique email list server with a one hour post-to-message lag time. RE performance, see above. As for the rest, it's opinion, not fact. For fun, you might register with www.vbulletin.com/forum and suggest they convert their entire community to an SMTP email list server and see how many people agree with you (generic you, not personal you). Kind of a tainted audiance, don't you think? Kinda like going to a sports bar and trying to convice people that being gay is the best thing for them. See how many converts you get. Please post the URL of the discussion where all the developers agree with you have much better vBulletin would be if they stopped building on-line communities and became a helter-skelter email-based .. Mess! The productivity of www.vbulletin.com and www.vbulletin.org surpasses the productivity and efficiency of this list by orders of magnitude (hands down). Just look at their archives, their posts, their announces, bug tracks, security releases, commercial support, etc. an infinitum. Again, subjective. I think Asterisk is doing very well thankyouverymuch. If you community is designed to pander to technical neophytes, it's going to work well for those neophytes. Open your eyes (them from the excellent movie Vanilla Sky)... ... And use an email client that works well with mailing lists!!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Fri, Apr 01, 2005 at 11:17:45AM -0600, Mike Hammett said: Ya, I mean do you really think an open source community is gonna acknowledge that MS can do anything right? of course not. THEY'RE THE DEVIL! (note, I will not respond to anything posted in reply to this, so don't even try) Fine. Insult the entire open source community (while using their work) and then run and hide. That shows maturity. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
On Sat, Mar 26, 2005 at 03:50:13AM -0500, Brian Capouch said: Asterisk could do the high-quality voice if it didn't care about interoperability. ... And things like echo supression. Skype doesn't do that *at all*. I fact, if you try to do a conference call with a remote user using skype to a group around a conference table (speaker phone mode), it's unusable. Lag can also be Really bad - several seconds - the worst of any VoIP system I've used. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
First, search the archives for skype. This question has come up before. Second, learn what skype is and how it works. Ditto for asterisk. See the respective websites and read the faqs there... The two are COMPLETELY different. One is a software pbx. The other is not. It's like comparing a car to a banana. On Sat, Mar 26, 2005 at 02:38:30AM +0800, Stephen said: Hi All, I face some problems when I try to introduce Asterisk to my customers / friends. They are not convince as they are currently using Skype and asking me what is/are the different between this two. Does anyone in the community can provide such a comparison chart? What's your opinion ? Thanks and Regards, Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Thu, Mar 10, 2005 at 01:04:35AM -0700, Forrest W. Christian said: I understand that PostgreSql has also gotten faster than it used to be. It's interesting. Just yesterday I was saying that we use both MySQL and Postgres here, and that we were probably going to move everything to postgres just to consolidate. Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. In the end, it all comes down to knowing exactly what features you need for your app, how your specific app performs on each DB, what you need for support, etc. As Forrest mentioned, write DB independant code and then you can easily choose the DB that is best for your app. 2 years for now, you may find a need to switch DB's. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Thu, Mar 10, 2005 at 08:55:53AM -0500, David Filion said: Walt Reed wrote: Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. Out of curiosity, what version of PostgreSQL was used? 7.x, 8.x? Also, was the test run on the same system? I'm not looking to bash. I'm just curious as we are in the same MySQL/PostgreSQL boat. We are useing 7.4.6. Considering 8.0 just came out in January, and considering how many major changes went into it, we were leary of upgrading until it had time to get tested by the masses. I would expect 8.x to be faster that 7.x, but I didn't see anything in the release notes that would indicate a 1 to 2 orders of magnitude performance increase. The tests were run on the same server (RHEL3 on a maxed out DL380-g4). We had been tuning the table design / query design, postgres config, etc. for quite some time, trying to get better performance. the mysql install was just the standard binaries available on the mysql site, with the default config. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: Best DB
On Thu, Mar 10, 2005 at 04:39:25PM +0100, Tom Ivar Helbekkmo said: Walt Reed [EMAIL PROTECTED] writes: I would expect 8.x to be faster that 7.x, but I didn't see anything in the release notes that would indicate a 1 to 2 orders of magnitude performance increase. A few points concerning PostgreSQL and performance: - Each of the latest releases has improved performance quite a bit. - Out of the box, it is tuned for minimal resource use, and dismal performance. It really needs to be tuned. Check out Josh Berkus's web site, http://www.powerpostgresql.com/, for hints and tips. - Nothing helps much if your schema and your queries are suboptimal. Think about how your data is used, consider what indexes you need, rewrite slow queries to be smarter (use EXPLAIN). - Did I mention you need to tune the database system to your needs? You snipped out my paragraph where I mentioned tuning the DB itself, queries, and schema. I have no doubt that 8.x is faster than 7.x, but I did not find any reports from people claiming a 10X performance boost. I didn't look hard, but I did look. I'll look into installing 8.x and see if we can rerun the tests. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Thu, Mar 10, 2005 at 09:09:09AM -0600, [EMAIL PROTECTED] said: I'd *love* to see the particulars of that test. It's been shown time and time again that postgres' speed CLOBBERS mysql for anything but the simplest selects, and that it can handle far more concurrent connections without slowing down. I strongly agree with this, i have a prepaid voip solution with asterisk, freeradius and postgresql , the hole thing relies in stored procedures and triggers (i mean the billing, traffic monitoring, admin system, etc). It had scaled from thousands of minutes per month to two millions in these days without an issue, we export the cdr to mysql for the IT/Customer Service OK, as some of you suspected, I found out that the test was serial. I'm having the programmer re-do the testing to be more representative of real-life - many concurent connections doing many different kinds of queries / inserts / updates at the same time. I too prefer postgres, but it's damn hard to state your case when someone hands you test results that show mysql beating the pants off it. I expect that we will see very different results under the new test. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 Phantom Ringing
On Tue, Mar 08, 2005 at 09:05:34PM -0500, Ben Ruset said: Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop ringing. Then, sometime later, it will happen on another random extension. Is this a common problem? Where can I look to start diagnosing this? You can run ethereal to capture all packets to / from the phone. Something is obviously causing this problem. If nothing shows up in ethereal, maybe there is a power problem. Are your phones POE or wall-wart? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
On Wed, Mar 09, 2005 at 03:02:03PM -0600, Steven Critchfield said: On Wed, 2005-03-09 at 15:43 -0500, [EMAIL PROTECTED] wrote: For some reason I didn't think PostgreSQL was for mission critical apps. I don't think I have any reasoning behind it, just didn't think it was hardcore...sounds like i might be wrong...i'll have to look into it more. Open source advantages are obvious, but aside from licensing and cost factors, I believe speed, security, and stability are going to be the key factors for us, whether open source or not. Postgres is probably more developed than mysql. Mysql gets a lot of press though as being an easy to install and config database. As for stability/scalability, the .org registry is on postgres. We use both MySQL and Postgres inhouse for production applications. MySQL has advanced significantly in the past year, and functionality is catching up with Postgres. Postgres is improving performance and is catching up to MySQL. Both are rock solid. Both have features the other lacks. I'd probably go with Postgres however for a new application. I see no point in giving MS or Oracle any more money for something that is a freely available commodity at this point. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block anonymous calls
On Sat, Mar 05, 2005 at 03:57:07PM -0600, Blake Van Eekeren said:
Fredrik wrote:
I see from my CDR's that some of my callers also have unknown in
their FROM field. I would like to let them through. Only block the
FROM anonymous that the telemarketers use.
Fredrik, I found something on the Wiki a while back... Try this...
exten = s,1,Answer
exten = s,2,NoOp(${CALLERID})
exten = s,3,ResponseTimeout(10)
exten = s,4,GotoIf($[${CALLERIDNUM} = ]?|1000)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|1000)
exten = s,6,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|1000)
exten = s,7,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|1000)
exten = s,8,Macro(stdexten,${SIP0})
exten = s,9,Hangup
exten = s,1000,Background(SPAMSTOPPER)
exten = s,1001,Hangup
Yeah, I put something like that on the wiki.
It works fairly well, but does not differentiate between anonymous and
unknown. This issue has come up several times on this mailing list and I
have yet to see a real solution.
I found that most telemarketers use unknown and not anonymous
actaully.
I require all calls without callerID to press 5 to get through. There
is also a privacy manager app that requests callers to enter their
number, but I feel that it's too annoying to friends / relatives.
I would rather have a special message for anonymous that is different. I
don't want those calls at all.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error connecting to remote mysql database.
On Wed, Feb 23, 2005 at 04:21:04AM -0800, R A said: I have this error when i try to conect to my remote mysql server: Host xxx.xxx.xxx.xxx is not allow to connect to this MySQL server. can some bady tell me what i have to do??? This has nothing to do with Asterisk. The error message tells you the problem. The manual tells you the solution. See: http://dev.mysql.com/doc/mysql/en/access-denied.html and: http://dev.mysql.com/doc/mysql/en/adding-users.html If you need further assistance, please use the mysql users mailing list. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.cc and/or Sixtel.net seems like IT IS A SCAM.
On Tue, Feb 15, 2005 at 07:42:56PM -0500, Nabeel Jafferali said: 4. Scnet.net has 5 pages website (quite a work for ISP), that any kid could create in 1h scnet.net is Server Central, a data centre where my host (HostForWeb), among others, maintains their servers. I do know it is a reliable data centre and I doubt is in any way related to iax.cc and/or sixtel.net other than housing their server(s). Yep. I've used server central as well. They DO have a reliable network, and hosting centers in several cities (including San Jose at Equinix.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redirect different phone number to different IP phone
On Fri, Jan 28, 2005 at 09:25:55AM -0500, Andrew Thompson said: Video Dery / Internet du Royaume wrote: Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers What kind of line? There has been some questions in the last day or so about DNIS, so I'm not sure that it can be done on inbound analog lines. I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 exten=5551234,1,Dial(SIP/phone1) exten=5551235,1,Dial(SIP/phone2) Customize accordingly... If on analog, you may be able to use distinctive ringing (zapata.conf) There are examples in the config file. Note that I think the dring code is limited to one zap interface which is unfortunate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom Phones
On Wed, Jan 26, 2005 at 10:20:24PM -0500, Cory Andrews said: Seshu - the 480i, although a great phones, is quite a bit more expensive than the Polycom IP300 or IP500, it is more comparable in price to the Polycom IP600. Hmm. Your own web site has it priced between the 500 and 600. If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one company tests with * and the other shuns it. Just the availability of the firmware alone is almost worth the $50. I have no problem with polycom, and use their non-IP conference phones, but I'm not going to purchase a product from a manufacturer that refuses to provide even basic support (complete manuals and firmware.) It would be Very nice to have a phone platform that is fully documented that had firmware that was open and hackable. It seems that people on this list spend massive amounts of time trying to work around all the firmware bugs in various products (eg. call waiting on polycom.) If sayson provided developer documentation for their phones and allowed us to write our own firmware, they wouldn't be able to manufacturer them fast enough. They would corner the IP phone market. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom phones
On Thu, Jan 27, 2005 at 10:45:53AM -0500, J Thomas said: If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one company tests with * and the other shuns it. I have the same dilemma with Polycom phones. Given their support (actually complete lack of), I am quite loathe to giving them business. On the other hand they are so darned cheap compared to other similar phones, I sure get tempted to use them if I can find a workaround. How do you get a workaround for lack of support? Working around one configuration problem this time is one thing, but what about next time? And the time after that? The support cost is not just initial install. You will need to work with your customer for years with this phone system, right? $220. If it were a matter of 1 or 2 phones, I will gladly go with SNOM or Sayson, but if I have to buy 50, Polycoms become irresistible. True. It's a no-brainer with small volume, but for installations that require that you support the product, it should be even more of a no-brainer. As a dealer / installer, how can you possibly sell a solution to a client where you have zero backup from the manufacturer? Don't you have contractual liability? Unlike Asterisk, you can't fix firmware problems yourself. Just more food for thought. It sure would be nice if Polycom removed their head from their ***. Our choices in good quality phones are slim. Our choices for Good quality phones with support are worse. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom phones
On Thu, Jan 27, 2005 at 11:30:26AM -0500, Kanuri, Seshu (Company IT) said: My opinion (guess) on Polycom's Asterisk policy is - It is not that Polycom does not want their phones to be used with Asterisk. At the price these phones are sold, they will not be able provide support for all the features (AKA bugs or quirks) of Asterisk and make them transparent to Asterisk SIP stack and more notably - be user friendly for the Asterisk newbie user community. :) That does not excuse them from not making the firmware or ducumentation available. There is no reason for them to not allow downloads or provide documentation - even requiring registration before download would be OK. Furthermore, one of the current issue people have (not being able to disable call-waiting) is going to be a problem for ANY sip PBX software, not just asterisk. If they had ONE internal advocate that monitored this list for 2 hours a day and provided feedback to internal engineering / product management, and *occasionally* provided information to the list on major issues people have, they could sell a LOT more of these phones and we would not be having this discussion. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Call-Waiting
On Thu, Jan 20, 2005 at 01:16:42PM +1100, Adam Goryachev said: On Wed, 2005-01-19 at 10:43 -0500, C F wrote: On Wed, 19 Jan 2005 15:44:44 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: On Tue, 2005-01-18 at 20:03 -0600, Eric Rees wrote: Has anyone been able to find a way to disable call-waiting on Polycom phones? I've not yet found any solution to this, and I haven't seen anyone else who has. Definitely please let us all know if you do find the answer... http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup Fixing lazy top-posting. Good call (bad pun intended... :-) setgroup doesn't work in all cases. Consider that the user may be receiving calls from methods other than the dialplan (eg, queues) I haven't thought it through, but I'll throw this idea into the wind... If you route all calls through an extension macro (inbound and outbound,) could you have an asterisk DB variable that is set/reset when a line is in use? I take it ChanIsAvail will return true if one call is already in progress which is why we can't use it... In addition, this call macro could add / remove extensions from a queue when a call is in progress... I have no idea what the impact would be if you did something like transfer a call... Sure would be nice if Polycom pulled their head out of their *$$ and started supporting their product properly. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External fax modem takeover of fxo?
Given all the issues with the fax DSP code and reliability, I'm looking into the option of using an external fax modem instead of trying to shoehorn asterisk into handling faxes properly... I'm thinking of hooking up the faxmodem to the line directly, then hooking the X100P to the modem's Phone port. Can I enable fax detection and use the fax extension to spawn an external command (halafax) to receive the fax? In theory, this should cause the modem to take over the line, disconnecting the X100P (causing a Red alarm) and receive the fax. What is the state of fax detection with the stable branch of * on an X100P? Does it work reliably? Do I need to wait() or will Dial still be able to detect faxes: exten = s,1,Answer exten = s,2,Dial(blah extension for phone|20|d) exten = fax,1,System(tell-halafax-to-answer) or is this required: exten = s,1,Answer exten = s,2,wait(5) exten = s,3,Dial(blah extension for phone|20|d) exten = fax,1,System(tell-halafax-to-answer) Anyone doing something like this already? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External fax modem takeover of fxo?
On Tue, Jan 18, 2005 at 09:40:47AM -0800, Lee Howard said: On 2005.01.18 08:36 Walt Reed wrote: I'm thinking of hooking up the faxmodem to the line directly, then hooking the X100P to the modem's Phone port. From what I've been able to determine, Asterisk would be most unhappy if you did that. On X100Ps that phone jack/port needs to simply be ignored. I have a telephone connected to mine, but it's only there in the off-chance of an emergency, power outage, or to troubleshoot problems with the FXO. I understand that. That's not what I am thinking of. I want the plug the modem in the the PSTN, and the X100P into the modem's Phone jack - NOT plug the modem into the X100P Phone jack. When my hypothetical System() call runs, the modem would pick up the phone and in the process disable the phone port on the fax modem which should throw a red-alarm on the X100P. When the fax is done, the fax modem would hangup and reconnect the X100P to the PSTN. Can I enable fax detection and use the fax extension to spawn an external command (halafax) to receive the fax? Uh, well, in the obvious way yes, but not in the way that you seem to be wanting to hear. exten = s,1,Answer exten = s,2,wait(5) exten = s,3, exten = fax,1,Dial(FXS_PORT) Where FXS_PORT is an FXS port where a HylaFAX-controlled modem is connected... or any other fax device for that matter. The problem is, though, that there seems to be a significant amount of data loss when receiving with this configuration... even using ULAW. Now, an ECM-aware fax device may compensate well-enough, but there's still a problem. Yep - I read about all those problems. I have heard very few sucess stories of running fax through * at all, which is why I was trying to find a way to avoid it. If I can get * to do the fax detection only and then kick off the external process that should be enough... What is the state of fax detection with the stable branch of * on an X100P? Does it work reliably? Fax detection is generally okay, but there are some problems. See: http://bugs.digium.com/bug_view_page.php?bug_id=0002165 Hmm. Interesting. So the problem is that Fax detection doesn't work, so the reporter goes through all the hoops and captures audio showing an exact case where detection is failing and therefore the bug is closed. He adds additional detail and the bug is closed AGAIN. Cute. So the real answer is that * can't handle fax correctly and reliably at all, so get a dedicated PSTN line... Is it just me or is the fact that * can't do what a $70 fax machine can do seem a little bizzare? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-1001 and Tivo Series 1
On Thu, Jan 06, 2005 at 11:50:40PM -0500, David Ishmael said: What about when users switch to 100% VoIP? I've been considering getting DirecTV with the HD PVR and I've heard it can't use broadband, in a case like that I would have to route a modem call through VoIP (or is there a better way I'm just not seeing). I've thought about this a little... It would be interesting to see if you could setup an spa2000 with a dialplan that calls another modem on the second port, and fake the PPP session. Maybe, just maybe, with the call being local to the device you can get it to work. Or some kind of T38 type solution... (BTW, your mail client doesn't quote properly. If running outlook, you can install quotefix to fix it.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kind of urgent
On Thu, Jan 06, 2005 at 01:08:06PM -0500, Gary G. Hendershot said: Try WhiteBox Linux ... It's a freeware clone of Redhat Enterprise Linux ... Its available for download in ISO CD image form (3 CD's required) ... Installs and configures just like Redhat ... Am using it now with SATA drives ... Works well with Asterisk ... Should be able to find a download site easily by Googling WhiteBox Linux ... Only problem with Whitebox is that it's one guy maintaining it, and updates are not recent. CentOS is another good alternative of you want to stick with the RHEL line. Much larger support community. FYI, I had been running WB on a couple servers and upgraded to CentOS on the fly with Yum (instructions are available on how to do this - google is your friend :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kind of urgent
On Thu, Jan 06, 2005 at 01:35:16PM -0500, Leif Madsen said: On Thu, 6 Jan 2005 19:32:24 +0200, Shoval Tomer [EMAIL PROTECTED] wrote: Can anyone comment why shouldn't we use FC 3 for an * production system? I'm not looking to start a distro war, but we just found out that redhat 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. We are only familiar with red hat and are in a point in time that switching distros is not available. The guy installing the system is already on location. Yes, I know we made a silly mistake. Please help us... Thanks. I'm not too sure what you want help with. You say that you are stuck to using FC3 because earlier versions don't support SATA drives? Could you use an older version and recompile the SATA drive support into the kernel? What about FC2 as it's 2.6 kernel based? Another option is installing a good SATA controller like the 3ware. Drives show up as SCSI, and even fairly old distro's work out of the box with them. Most onboard controllers are pretty crappy - avoid Promise and their fake RAID. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kind of urgent
On Thu, Jan 06, 2005 at 12:03:32PM -0700, Michael Welter said: Walt Reed wrote: Another option is installing a good SATA controller like the 3ware. Drives show up as SCSI, and even fairly old distro's work out of the box with them. Most onboard controllers are pretty crappy - avoid Promise and their fake RAID. Don't the SATA drives also show-up as SCSI devices? I don't recall having any problems with SATA drives. It's not the drives, it's the controller and the drivers that are the big issue. When the linux drivers talk to a normal SATA controller, they don't see SCSI drives out there, but they present them as SCSI to the next layer in the OS. 3ware controllers have a unified driver model. Raidsets show up as scsi drives to the linux driver, no matter which card and drive technology you use (IDE, or SATA.) Promise controllers on the other hand seem to need totally new drivers with each sub revision. Bah! Most SATA controllers need libata (in 2.4) to work. 3ware controllers do not. Hope this helps someone... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qs about FXO/FXS cards
On Mon, Jan 03, 2005 at 12:27:56PM -0500, Andrew Kohlsmith said: My Panasonic 900MHz cordless phone plays silly bugger with the TDM400P card all the time. For whatever reason it either draws far too much power or just plain does not like the TDM430P. My Aastra 390 and a couple other regular phones seem to work just fine, but that cordless phone will crackle and sputter for the first 10s or so of the call, at which point it quiets down and behaves itself. It's interesting - in addition to my own experiences, I have heard Many stories about Panasonic cordless phones acting strange / causing problems on all sorts of phone systems - not just asterisk. Enough so, that I tell people to avoid them. It would be interesting to see exactly what they do to the line that makes them more problematic that other brands. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telemarketer screening
On Tue, Aug 24, 2004 at 03:34:34PM +1000, david kwok said: I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? http://lists.digium.com/pipermail/asterisk-users/2004-August/059836.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list broken again?
On Tue, Dec 21, 2004 at 11:05:27PM -0500, Alex Brecher said: I still don't get why we don't move over to a web based forum ? Because web-based forums suck. The only people who seem to like web forums are those that don't know how to use their email client (or use a client so braindead that it doesn't even do threading.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Old posts and the ability to search...
On Fri, Dec 17, 2004 at 04:53:27PM -, Paul Brock said: Just a passing thought... is there any reason why the ability to search the past posts on here isn't switched on? Just wondered, since it makes much more sense to be able to search the old archives if you have a problem, rather than ask the same question again and again... Use google with the Site: option. ___ 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] using built-in extension numbers on the ZAP channel
On Sat, Dec 11, 2004 at 12:05:02AM +0600, Samudra E. Haque said: hello, using a legacy PBX to access a Asterisk Zap channel (Legacy PBX FXS -- FXO application Asterisk/TDM400P) I want to be able to flash the asterisk pbx. However by pressing the FLASH button on the extension connected to the Legacy PBX gets me the flash features on the Legacy PBX, not on the Asterisk PBX side. I thought of using the following codes (listed below) from http://www.voip-info.org/wiki-Asterisk+zap+channels but, when i dialled from the extension (legacy pbx) -- extension of Asterisk pbx - zaptel/zap channel - IVR, and pressed *0, it was invalid extension. How can I pass on a 'flash' key / command so that I can flash the remote side instead of the local side ? *0 will send a flash from a * FXS through to an FXO: Phone presses *0 - FXS on * - * - flash FXO - PSTN receives flash This is not what you want / need. You need to configure your legacy PBX to send a flash through from your legacy phone to the legacy FXS port that is hooked to a * FXO port. This is not something you can do from the * side. *0 is asterisk's way of doing this. You need to find out if your legacy PBX has a similar method. It may not be possible. Of course you failed to mention what kind of PBX you have, so nobody can possibly advise you further, and at that point it's off-topic. ___ 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] Ethernet Channel Bank idea
On Wed, Dec 08, 2004 at 08:43:10PM -0600, nik martin said: Anyone ever thought about an Ethernet based channel bank? Basically a rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with zaptel, etc. IAX as the * - Channel bank protocol. Yes. Search the list :-) My idea was close, but a little different. I'd like to see an ethernet unit that used the same modules as the TDM400P. Could even be a couple of models: a 6 port, 12 port, and 24 port. Don't know about the 24 port unit as a channel bank eats into that market. The 6 and 12 port modules would be perfect for soho and small business that don't need the port density of a full blown channel bank. This is the market that Asterisk is AWESOME for. Stick * on a small embedded linux box like a linksys router and plug in a 6-port IAXy with 4 FXS and 2 FXO. Need to expand? Add another 6 port IAXy. The modules for the TDM400P are economical enough to do this with and beat out pricing of most of the larger (more than 2 port) SIP gateways AND already have certification (or pending certification.) Of course the big bonus is IAX instead of SIP. Frankly, with all the interupt issues with PCI, especially if you need more than one card, a low-port density alternative to the TDM400P is needed. This should really increase the volume of Digium's sales as there would be a better alternative to the SPA-2000 and 3000. Increased volum drives down the cost of the modules (which if they were made in china should drop to ~10-20 each.) Another reason for going small (6 port) is that you can use a embedded low-power inexpensive processor for codec processing and keep it inexpensive. ___ 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 Maintenance
On Wed, Dec 08, 2004 at 01:44:10PM -0700, Michael Welter said: I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why would the LEDs go off? Why would keyboard input fail? I've seen cases where the keyboard totally locks up - sometimes hardware glitch, sometimes software. In the hardware cases, you can generally unplug / re-plug the keyboard and wake things up again. Long-shot, but maybe static or power surge screwed things up? ___ 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] drive space for voice mail
On Thu, Dec 02, 2004 at 10:18:56PM -0600, Steven Critchfield said: On Thu, 2004-12-02 at 21:08 -0600, [EMAIL PROTECTED] wrote: Thank you everyone for your input... I think I'm safe at reducing them to 2 X 80GB (RAID 1) and still have plenty of room (these drives will also include the OS, * install etc). Eventually the server may end up servicing many remote sites, so I don't mind being slightly over the top. Maybe you should research the problems of your raid solution. http://www.acnc.com/04_01_01.html IDE raid 1 should be avoided. IDE drives themselves can cause degraded performance on a machine and raid 1 would double the IDE activity. If you were to use a hardware raid option, you would reduce the likely hood of degraded performance due to IDE activity. Use a good card like the 3ware 7500 series (parallel IDE ATA) and there are no problems using IDE ATA drives. 3ware uses hardware raid unlike the garbage promise chips that Claim hardware raid, but are not in reality. IED Raidsets on 3ware show up as scsi drives to the system. 3ware is one of those rare companies that have Great linux support. You get what you pay for. The controller card may cost as much or more than the drives. Linux SATA support is still a little weak, but the performance can be much better for the higher-end SATA drives. Use of a good raid card like 3ware makes Linux compatability a non-issue. I agree that software raid should be avoided. ___ 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
