Re: [Asterisk-Users] Cisco 7960 SIP Images
Hi Chris, Chris HARIGA wrote: If you pay 8 USD for 1 year support you can download the image :) Best regards, Someone mentioned a while ago that even if you have a support contract for the SIP image this doesn't mean you have a license to use the software. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax/spandsp: file-naming of received faxes
Hi,
after successfully having installed RxFax/SpanDSP and some promising tests
(great piece of software, Steve!) I wonder if it is possible to avoid
overwriting the same tiff file over and over again.
Browsing the sourcecode of app_rxfax.c I found a magic '%d' flag being parsed
out from the argument of rxfax(), but didn't manage to make that work.
extensions.conf:
exten = _3XX,1,rxfax(/tmp/faxfor-${EXTEN}-%d.tif)
always produces filenames like 'faxfor-345-0.tif', so the %d is handled somehow.
My intension is to watch the fax directory for new files by a cron job and email
them to their recipients.
It would be nice to have %d being replaced by a timestamp like 20040328120208 or
simply an incrementing integer and if possible have the calling station id added
to the filename.
Any hints are greatly appreciated.
Thanks a lot,
Jan
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[Asterisk-Users] Registers
How many registers SIP I can place in the Asterisk? Thank's Jmoura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoftFAX/spandsp
Hi all, spandsp-0.0.1h.tar.gz seemed to get a lot more FAX machines talking, but a number of people are getting rather high error rates on the images. spandsp-0.0.1i.tar.gz addresses this, and should give much better bit error rates for fax machines whose timing at the extremes the spec allows. It can be found at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1i.tar.gz Regards, Steve ___ 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] SoftFAX/spandsp
Steve Underwood wrote: Hi all, spandsp-0.0.1h.tar.gz seemed to get a lot more FAX machines talking, but a number of people are getting rather high error rates on the images. spandsp-0.0.1i.tar.gz addresses this, and should give much better bit error rates for fax machines whose timing at the extremes the spec allows. It can be found at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1i.tar.gz Regards, Steve Don't try using libtiff 3.6.1 with spandsp, or HylaFAX, or any other fax package. It has bugs in its FAX image handling. Regards, Steve ___ 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] Registers
Joao Carlos Moura [EMAIL PROTECTED] wrote: How many registers SIP I can place in the Asterisk? Thank's There's no limit. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
Hi Jan,
Try this:
exten = _3XX,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten = _3XX,2,rxfax(${FAXFILE})
Good luck,
- Original Message -
From: Jan Baumann [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 7:09 AM
Subject: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
after successfully having installed RxFax/SpanDSP and some promising tests
(great piece of software, Steve!) I wonder if it is possible to avoid
overwriting the same tiff file over and over again.
Browsing the sourcecode of app_rxfax.c I found a magic '%d' flag being
parsed
out from the argument of rxfax(), but didn't manage to make that work.
extensions.conf:
exten = _3XX,1,rxfax(/tmp/faxfor-${EXTEN}-%d.tif)
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RE: [Asterisk-Users] Codec Voodoo: piece of evidence: probable fix
FWIW, the fix below applies to cvs (not Stable). The stable version of this
is very different. Looks like Stable hasn't been updated for some time.
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
rtp-txcore.tv_usec -= rtp-txcore.tv_usec % 2;
}
if (delivery (delivery-tv_sec || delivery-tv_usec)) {
/* Use previous txcore */
=ms = (delivery-tv_sec - rtp-txcore.tv_sec) * 1000;
ms += ((delivery-tv_usec - rtp-txcore.tv_usec) + 500) /
1000;
rtp-txcore.tv_sec = delivery-tv_sec;
rtp-txcore.tv_usec = delivery-tv_usec;
} else {
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
=ms += ((now.tv_usec - rtp-txcore.tv_usec) + 500 ) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
}
return ms;
}
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Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
working thank you
Nicolas Gudino wrote:
Hi Jan,
Try this:
exten = _3XX,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten = _3XX,2,rxfax(${FAXFILE})
Good luck,
- Original Message -
From: Jan Baumann [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 7:09 AM
Subject: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
after successfully having installed RxFax/SpanDSP and some promising tests
(great piece of software, Steve!) I wonder if it is possible to avoid
overwriting the same tiff file over and over again.
Browsing the sourcecode of app_rxfax.c I found a magic '%d' flag being
parsed
out from the argument of rxfax(), but didn't manage to make that work.
extensions.conf:
exten = _3XX,1,rxfax(/tmp/faxfor-${EXTEN}-%d.tif)
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Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
On Mar 28, 2004, at 5:21 AM, Nicolas Gudino wrote:
Hi Jan,
Try this:
exten = _3XX,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten = _3XX,2,rxfax(${FAXFILE})
Or
exten =
_3XX,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
If you want a unique id, why not use the one that Asterisk provides?
In case anyone's interested, I spent a bit of time on incoming faxes
yesterday, prototyping a DID FAX-type setup. Here are a few snippets,
in case anyone's interested.
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
[fax]
exten = 2201,1,Macro(faxreceive)
exten = 2202,1,Macro(faxreceive)
exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} \
${CALLERIDNUM} ${CALLERIDNAME})
; I'm using a shared analog line for testing this, so I'm using the fax
; autodetection code to yank faxes out of my IVR and into the 'fax'
; pseudo-extension
[outside]
...
exten = fax,1,Goto(fax,2201,1)
Finally, here's /usr/local/sbin/mailfax:
#!/bin/sh
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
tiff2ps -2eaz -w 8.5 -h 11 $FAXFILE |
ps2pdf - |
mime-construct --to $RECIPIENT --subject Fax from $FAXSENDER
--attachment fax.pdf --type application/pdf --file -
In Debian, tiff2ps comes in libtiff-tools, ps2pdf is part of
Ghostscript, and mime-construct is its own package.
To set the email address associated with each extension, do 'database
put extensionemail EXTEN [EMAIL PROTECTED]'
Scott
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[Asterisk-Users] OT Sipura: Sipura doesn't see * hangup PSTN line
I have an Siemens two-line base station connected as follows: Line 1: Sipura SPA-2000 Line 1 Line 2: Sipura SPA-2000 Line 2 I have a single POTS line (Verizon) connected to a Digium X100P card. [Config excerpts are below] Issue: Calling into the * server rings the Siemens base station as expected. After about 20 seconds the Siemens base station answers the call and sends the call to voicemail (I know I should use VM in * but my spouse isn't yet ready for that...). == Spawn extension (from-verizon, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Then the caller hangs up. Asterisk sees this hangup and disconnects the PSTN line (I confirm this with a monitor phone on the POTS line). Asterisk sends the SIP BYE command, and receives the OK response from the Sipura. For some reason the Sipura doesn't hangup line 1 and, as a result, every message on the Siemens base station has a ~15 second Congestion tone trailer. Any ideas? Most likely the answer is bug in Sipura, move quickly to *-based VM, right? :^) Relevant zapata.conf entry: - [channels] language=en context=from-verizon signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 Relevant extensions.conf entry: - [from-verizon] exten = s,1,Dial(SIP/sipura_line1) exten = s,2,Hangup exten = h,1,Hangup exten = i,1,Hangup Relevant sip.conf entry: - [sipura_line1] type=friend username=sipura_line1 secret=XX host=dynamic context=from-siemens-line-1 mailbox=9000 canreinvite=no callerid = 8885551212 -- Rob PageV: 540.361.1710 Zope CorporationF: 703.995.0412 ___ 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] Cisco 7960 SIP Images
I have recieved far more that my money's worth in technical calls to Cisco about my 7960 telephones. They respond immediately. They keep working until the job is done. The pull in whatever resources are neccessary. They have never failed to find and fix the problem. If you want professional, real technical support you should be willing to pay for it, or in this case part of it. Paul Mahler mailto:[EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 27, 2004 7:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP Images What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware companies have any VoIP phones worth a crap? Then deal with the fact that you are buying from a company who doesn't target home users, and deal with it. It costs Cisco more money than they make on the contract to offer SmartNet on a single device like this. You're lucky they don't have a minimum device limit/contract cost of something like 5 devices or $300/year. I'm guessing this type of policy would hardly effect more than several hundred of their customers, most of them with 7960's and similar. -Original Message- From: [EMAIL PROTECTED] on behalf of John Baker Sent: Sat 3/27/2004 4:41 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images [massive amounts trimmed] No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat
[Asterisk-Users] two-stage dialing
I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. Bridge channel with calling party I thought that something like: exten = _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten = _2XX,3,Wait,1 exten = _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov ___ 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] Cisco 7960 SIP Images
I got a 7960 for evaluation purposes. I was planning on upgrading our phone system and wanted to see if Cisco's product was any good. Short answer: Nice phone, horrible service. Support? I called Cisco looking for support on the phones. They told me to go through a reseller, and I could find one on their website. I contacted a local reseller, listed as having the Cisco line on Cisco's website and guess what - they had no idea what I was talking about. Seems they didn't even know they were listed on Cisco's website to begin with. I tried a second reseller with similar results. I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John Paul Mahler wrote: I have recieved far more that my money's worth in technical calls to Cisco about my 7960 telephones. They respond immediately. They keep working until the job is done. The pull in whatever resources are neccessary. They have never failed to find and fix the problem. If you want professional, real technical support you should be willing to pay for it, or in this case part of it. Paul Mahler mailto:[EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 27, 2004 7:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP Images What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware companies have any VoIP phones worth a crap? Then deal with the fact that you are buying from a company who doesn't target home users, and deal with it. It costs Cisco more money than they make on the contract to offer SmartNet on a single device like this. You're lucky they don't have a minimum device limit/contract cost of something like 5 devices or $300/year. I'm guessing this type of policy would hardly effect more than several hundred of their customers, most of them with 7960's and similar. -Original Message- From: [EMAIL PROTECTED] on behalf of John Baker Sent: Sat 3/27/2004 4:41 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images [massive amounts trimmed] No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP Images
They do you just need a CCO and a Smartnet contract for your phone. Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Saturday, March 27, 2004 4:06 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images Welcome to the very much less than wonderful world of Cisco software support. When will those guys simply make the software downloadable straight away from their website for a modest fee? Iain --On Saturday, March 27, 2004 1:43 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: I just received my first Cisco 7960 today and was looking forward to playing with it this weekend, however I can't seem to get it working via skinny (can't find any information via the wiki regarding what needs to be on the tftp server for skinny). I would like to get my hands on the SIP images to play with it. I know I have to get a support contract through Cisco to get download access via their site which you can bet I'm going to do Monday morning, but I was hoping to work with it this weekend while I have the time. I found the release 4.4 SIP image, but it won't take due to a bug that was evidently fixed around v3.? (4k tftp buffer, and the new image is larger). At least I have a really expensive pretty phone sitting on my desk now! :-) Mitch Sharp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Images
John, I completely agree with you. I had the exact same problem you did just 2 years ago. It was our first time dealing with Cisco and I was so disappointed with their service and attitude. I guess as size and fame goes up for a company, service and friendliness goes down. May be some PHD should do a thesis on that. By the time the Polycom phones were available, we completely jumped on them!!! David - Original Message - From: John Baker [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 28, 2004 11:22 AM Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images I got a 7960 for evaluation purposes. I was planning on upgrading our phone system and wanted to see if Cisco's product was any good. Short answer: Nice phone, horrible service. Support? I called Cisco looking for support on the phones. They told me to go through a reseller, and I could find one on their website. I contacted a local reseller, listed as having the Cisco line on Cisco's website and guess what - they had no idea what I was talking about. Seems they didn't even know they were listed on Cisco's website to begin with. I tried a second reseller with similar results. I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John Paul Mahler wrote: I have recieved far more that my money's worth in technical calls to Cisco about my 7960 telephones. They respond immediately. They keep working until the job is done. The pull in whatever resources are neccessary. They have never failed to find and fix the problem. If you want professional, real technical support you should be willing to pay for it, or in this case part of it. Paul Mahler mailto:[EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 27, 2004 7:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP Images What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware companies have any VoIP phones worth a crap? Then deal with the fact that you are buying from a company who doesn't target home users, and deal with it. It costs Cisco more money than they make on the contract to offer SmartNet on a single device like this. You're lucky they don't have a minimum device limit/contract cost of something like 5 devices or $300/year. I'm guessing this type of policy would hardly effect more than several hundred of their customers, most of them with 7960's and similar. -Original Message- From: [EMAIL PROTECTED] on behalf of John Baker Sent: Sat 3/27/2004 4:41 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images [massive amounts trimmed] No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] SoftFAX/spandsp
Hi, -Original Message- spandsp-0.0.1h.tar.gz seemed to get a lot more FAX machines talking, but a number of people are getting rather high error rates on the images. spandsp-0.0.1i.tar.gz addresses this, and should give much better bit error rates for fax machines whose timing at the extremes the spec allows. It can be found at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1i.tar.gz Cool, I am really happy to see this stuff moving along. After a short time of not being able to track updates I tried getting back up there again, updated app_rxfax (whoof, that was a while back). Successfully built and installed spandsp 1i, and started asterisk (tonight's cvs): gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from /usr/local/include/spandsp.h:40, from app_rxfax.c:29: /usr/local/include/spandsp/arctan2.h: In function `arctan2': /usr/local/include/spandsp/arctan2.h:44: warning: type mismatch in implicit declaration for built-in function `fabs' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:185: too few arguments to function `ast_set_read_format' app_rxfax.c:195: too few arguments to function `ast_set_write_format' app_rxfax.c:199: too few arguments to function `ast_set_read_format' app_rxfax.c:247: too few arguments to function `ast_set_read_format' app_rxfax.c:253: too few arguments to function `ast_set_write_format' make[1]: *** [app_rxfax.o] Error 1 H - compilation aborts. Any thoughts ?? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] opaque missing in Authorization header
Title: opaque missing in Authorization header Hi folks, isn't opaque missing in this reply from asterisk? I'm using newest CVS code and newest partysip server (2.1.1 although 0.6.0 mentioned below). Is there a chance to configure that somehow instead of lookin in the code? But I guess there is no element in p- something for opaque in chan_sip.c? W. Sorry, if that has been answered already, but I searched for an answer the whole weekend and found none. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.152.191.65:5060;branch=z9hG4bK7f31ebcd From: sip:[EMAIL PROTECTED];tag=as3c8a81cd To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=labor.at, nonce=e7eef83591e1a19c0b5e8c5474e9c2b7, opaque=26da62c15c5cd1c2d008a68b472602e3 Server: partysip/0.6.0 Content-Length: 0 REGISTER sip:penelope.labor.at SIP/2.0 Via: SIP/2.0/UDP 192.152.191.65:5060;branch=z9hG4bK7f31ebcd From: sip:[EMAIL PROTECTED];tag=as3c8a81cd To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=asterisk, realm=labor.at, algorithm=MD5, uri=sip:penelope.labor.at, nonce=e7eef83591e1a19c0b5e8c5474e9c2b7, response=b1e479c2ad8eebc0c002113ee059f1c9 Expires: 1800 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0
RE: [Asterisk-Users] SoftFAX/spandsp
On Sun, 28 Mar 2004, Florian Overkamp wrote: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from /usr/local/include/spandsp.h:40, from app_rxfax.c:29: /usr/local/include/spandsp/arctan2.h: In function `arctan2': /usr/local/include/spandsp/arctan2.h:44: warning: type mismatch in implicit declaration for built-in function `fabs' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:185: too few arguments to function `ast_set_read_format' app_rxfax.c:195: too few arguments to function `ast_set_write_format' app_rxfax.c:199: too few arguments to function `ast_set_read_format' app_rxfax.c:247: too few arguments to function `ast_set_read_format' app_rxfax.c:253: too few arguments to function `ast_set_write_format' make[1]: *** [app_rxfax.o] Error 1 The ast_set_read_format and ast_set_write_format functions have been changed in CVS head (but not stable), to include a flag if the channel should be locked. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Michael's Minute: Two New Products - Call-in-One, Enterprise Assessment Kit
Title: Lindows.com Michael's Minute - Original Message - From: Michael Robertson To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 11:50 AM Subject: Michael's Minute: Two New Products - Call-in-One, Enterprise Assessment Kit If this message is not displaying properly, click here to launch it in your browser. Michael's Minute: Two New Products - Call-in-One, Enterprise Assessment Kit There are two new products available for the first time this week that I am excited about. One is from Lindows and the other from SIPphone (my other company). Call-in-OneStarting this week, a brilliant new device is making its worldwide debut at SIPphone.com which I'm predicting will make Internet calling practical for millions. Last year I started a company around a technology called SIP, which I'm convinced will revolutionize telecom. SIP gives consumers power over their personal telephone, much like MP3 gave them control of their music libraries. But the problem with SIP is that a special phone or a PC has been necessary to make these high-quality Internet calls. Nobody wants a second phone on their desk just for 'Net calls. Computers are great, but too heavyweight for the phone experience. Dealing with headsets and making sure the software is installed and running is bothersome. What people want is a way for their existing phones to receive and dial net calls. So last year I met with a company called LeadTek and asked them to build a device that I could plug in at my home that would allow me to do SIP calls. Most importantly, it had to be invisible to my wife. I wanted her to be able to make and receive SIP calls with no change to her existing phone experience. I wanted to be able to use the same cordless unit and dial numbers exactly as she is accustomed to. (You might know LeadTek from their popular line of GeForce graphics cards.) Get the Call-in-One from SIPphone.com LeadTek delivered a paperback novel-sized unit known as the "Call-in-One" to fit the bill. My wife suggested this clever name when I described how this device would permit the same phone in our house to make regular calls and free SIP calls. To use the Call-in-One, simply connect it to your broadband Internet connection and a line from the phone jack on wall. Then you plug the same phone you currently use into the Call-in-One box instead of the phone jack in the wall. Presto! You've added free worldwide calling to your phone. Because the Call-in-One uses plug-n-dial technology, the device is auto-configured and an available SIP number is assigned upon first use. After just a few seconds, you'll be ready to dial any SIP number in the world by first dialing the # key, or receive calls from more than 100,000 numbers on the SIPphone network. Of course, regular calls are dialed and answered exactly as before. The Call-in-One adds free worldwide SIP calling as a new feature to your phone and best of all, it does it an elegant and easy-to-use way. Anyone currently paying long distance bills - especially those who make international calls - can save a bundle by using a Call-in-One. Because calls are transmitted over the Internet there are no per minute or monthly fees for SIP-to-SIP calls. You will need to make sure that the people you call the most also have a SIP number, so you can maximize your savings. Every Call-in-One or SIP adapter comes with a free SIP number which will look something like 1-747-123-4567. In addition, my.sipphone is a free service provided to Call-in-One users, so they can manage their phone book, track call history and adjust features like voice mail all via email. Would you spend a one-time fee of $89 to buy a Call-in-One to add free unlimited worldwide 'Net calling to your existing home or business phone? The first manufacturing run of these units was only one thousand units, which are now in the SIPphone warehouse, so you'll want to get your order in quickly to be one of the first to get this phenomenal money-saving device. Enterprise Assessment KitThe other new product is from Lindows, and it demonstrates the increasing interest from enterprises for desktop Linux. Many are eager to determine how well Linux will work in their business or school, but are unsure where to start an evaluation process. The $149 Desktop Linux Enterprise Assessment Kit is a toolbox that allows you to experience state-of-the-art desktop Linux products, and survey the functionality and
[Asterisk-Users] OT - Error compiling screen
When I compile screen on my * server I get the following errors. Any pointers would be greatly appreciated. [EMAIL PROTECTED] screen-3.9.15]# make gcc -c -I. -I.-g -O2 screen.c In file included from screen.h:45, from screen.c:85: term.h:40:1: warning: d_CUP redefined term.h:38:1: warning: this is the location of the previous definition term.h:41:1: warning: D_CUP redefined term.h:39:1: warning: this is the location of the previous definition term.h:44:1: warning: d_CDO redefined term.h:42:1: warning: this is the location of the previous definition term.h:45:1: warning: D_CDO redefined term.h:43:1: warning: this is the location of the previous definition term.h:52:1: warning: d_CLE redefined term.h:50:1: warning: this is the location of the previous definition term.h:53:1: warning: D_CLE redefined term.h:51:1: warning: this is the location of the previous definition term.h:72:1: warning: d_CDL redefined term.h:70:1: warning: this is the location of the previous definition term.h:73:1: warning: D_CDL redefined term.h:71:1: warning: this is the location of the previous definition term.h:82:1: warning: d_CIC redefined term.h:80:1: warning: this is the location of the previous definition term.h:83:1: warning: D_CIC redefined term.h:81:1: warning: this is the location of the previous definition term.h:86:1: warning: d_CDC redefined term.h:84:1: warning: this is the location of the previous definition term.h:87:1: warning: D_CDC redefined term.h:85:1: warning: this is the location of the previous definition term.h:94:1: warning: d_CCD redefined term.h:92:1: warning: this is the location of the previous definition term.h:95:1: warning: D_CCD redefined term.h:93:1: warning: this is the location of the previous definition term.h:147:1: warning: d_CSF redefined term.h:62:1: warning: this is the location of the previous definition term.h:148:1: warning: D_CSF redefined term.h:63:1: warning: this is the location of the previous definition term.h:153:1: warning: d_CCO redefined term.h:24:1: warning: this is the location of the previous definition term.h:154:1: warning: D_CCO redefined term.h:25:1: warning: this is the location of the previous definition term.h:165:1: warning: d_CCS redefined term.h:58:1: warning: this is the location of the previous definition term.h:166:1: warning: D_CCS redefined term.h:59:1: warning: this is the location of the previous definition term.h:167:1: warning: d_CCE redefined term.h:96:1: warning: this is the location of the previous definition term.h:168:1: warning: D_CCE redefined term.h:97:1: warning: this is the location of the previous definition term.h:175:1: warning: d_CWS redefined term.h:110:1: warning: this is the location of the previous definition term.h:176:1: warning: D_CWS redefined term.h:111:1: warning: this is the location of the previous definition term.h:195:1: warning: d_COP redefined term.h:151:1: warning: this is the location of the previous definition term.h:196:1: warning: D_COP redefined term.h:152:1: warning: this is the location of the previous definition screen.c: In function `serv_select_fn': screen.c:2973: `D_VB' undeclared (first use in this function) screen.c:2973: (Each undeclared identifier is reported only once screen.c:2973: for each function it appears in.) make: *** [screen.o] Error 1 Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error installing/compiling cdr_mysql addon
When I try to compile the cdr_mysql addon, I get the following error: [EMAIL PROTECTED] asterisk-addons]# make cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I have MySQL installed and have tested it - it is working, I can create databases etc. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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] SoftFAX/spandsp
Hi James, James Golovich wrote: On Sun, 28 Mar 2004, Florian Overkamp wrote: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from /usr/local/include/spandsp.h:40, from app_rxfax.c:29: /usr/local/include/spandsp/arctan2.h: In function `arctan2': /usr/local/include/spandsp/arctan2.h:44: warning: type mismatch in implicit declaration for built-in function `fabs' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:185: too few arguments to function `ast_set_read_format' app_rxfax.c:195: too few arguments to function `ast_set_write_format' app_rxfax.c:199: too few arguments to function `ast_set_read_format' app_rxfax.c:247: too few arguments to function `ast_set_read_format' app_rxfax.c:253: too few arguments to function `ast_set_write_format' make[1]: *** [app_rxfax.o] Error 1 The ast_set_read_format and ast_set_write_format functions have been changed in CVS head (but not stable), to include a flag if the channel should be locked. James Locked in what sense? I am not clear whether I should set it to TRUE or FALSE. :-\ Regards, Steve ___ 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] Error installing/compiling cdr_mysql addon
You will need to install libmysqlclient10 Zac -- Zac Amsler, Technical Team WNOC.COM http://www.wnoc.com Phone: (801) 606-8047 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Sunday, March 28, 2004 6:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error installing/compiling cdr_mysql addon When I try to compile the cdr_mysql addon, I get the following error: [EMAIL PROTECTED] asterisk-addons]# make cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I have MySQL installed and have tested it - it is working, I can create databases etc. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error installing/compiling cdr_mysql addon
Simon Brown wrote: I have MySQL installed and have tested it - it is working, I can create databases etc. You need the dev headers/libs not just the main binaries... most distributions install these separately if you only need the mysql server hooked up to php binaries... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ 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] OT - Error compiling screen
You are obviously missing a dependency. I would search google for your solutions or use a distro that has auto dependencies. Zac -- Zac Amsler, Technical Team WNOC.COM http://www.wnoc.com Phone: (801) 606-8047 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Sunday, March 28, 2004 6:01 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT - Error compiling screen When I compile screen on my * server I get the following errors. Any pointers would be greatly appreciated. [EMAIL PROTECTED] screen-3.9.15]# make gcc -c -I. -I.-g -O2 screen.c In file included from screen.h:45, from screen.c:85: term.h:40:1: warning: d_CUP redefined term.h:38:1: warning: this is the location of the previous definition term.h:41:1: warning: D_CUP redefined term.h:39:1: warning: this is the location of the previous definition term.h:44:1: warning: d_CDO redefined term.h:42:1: warning: this is the location of the previous definition term.h:45:1: warning: D_CDO redefined term.h:43:1: warning: this is the location of the previous definition term.h:52:1: warning: d_CLE redefined term.h:50:1: warning: this is the location of the previous definition term.h:53:1: warning: D_CLE redefined term.h:51:1: warning: this is the location of the previous definition term.h:72:1: warning: d_CDL redefined term.h:70:1: warning: this is the location of the previous definition term.h:73:1: warning: D_CDL redefined term.h:71:1: warning: this is the location of the previous definition term.h:82:1: warning: d_CIC redefined term.h:80:1: warning: this is the location of the previous definition term.h:83:1: warning: D_CIC redefined term.h:81:1: warning: this is the location of the previous definition term.h:86:1: warning: d_CDC redefined term.h:84:1: warning: this is the location of the previous definition term.h:87:1: warning: D_CDC redefined term.h:85:1: warning: this is the location of the previous definition term.h:94:1: warning: d_CCD redefined term.h:92:1: warning: this is the location of the previous definition term.h:95:1: warning: D_CCD redefined term.h:93:1: warning: this is the location of the previous definition term.h:147:1: warning: d_CSF redefined term.h:62:1: warning: this is the location of the previous definition term.h:148:1: warning: D_CSF redefined term.h:63:1: warning: this is the location of the previous definition term.h:153:1: warning: d_CCO redefined term.h:24:1: warning: this is the location of the previous definition term.h:154:1: warning: D_CCO redefined term.h:25:1: warning: this is the location of the previous definition term.h:165:1: warning: d_CCS redefined term.h:58:1: warning: this is the location of the previous definition term.h:166:1: warning: D_CCS redefined term.h:59:1: warning: this is the location of the previous definition term.h:167:1: warning: d_CCE redefined term.h:96:1: warning: this is the location of the previous definition term.h:168:1: warning: D_CCE redefined term.h:97:1: warning: this is the location of the previous definition term.h:175:1: warning: d_CWS redefined term.h:110:1: warning: this is the location of the previous definition term.h:176:1: warning: D_CWS redefined term.h:111:1: warning: this is the location of the previous definition term.h:195:1: warning: d_COP redefined term.h:151:1: warning: this is the location of the previous definition term.h:196:1: warning: D_COP redefined term.h:152:1: warning: this is the location of the previous definition screen.c: In function `serv_select_fn': screen.c:2973: `D_VB' undeclared (first use in this function) screen.c:2973: (Each undeclared identifier is reported only once screen.c:2973: for each function it appears in.) make: *** [screen.o] Error 1 Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Images
I think John's said it all - I have absolutely nothing to add! I'm just posting to second his opinion. Terence On 29 Mar 04, at 3:22 AM, John Baker wrote: -- snip -- I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John ___ 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] Cisco 7960 SIP Images
I had a completely different experience. The day I decided I wanted to get a contract, I called Cisco, gave them my personal credit card, and three hours later had my CCO access upgraded. I just bought a smartnet for one phone for two years (a whopping $16), there was nothing to it. I've never been contacted by a sales rep (as a result of this purchase). I had an issue with the firmware not functioning properly - inside of two weeks, they had released a new firmware version resolving that problem and a few others. I'm not sure why the experiences would have been so different, but they are. Steve On Mon, 29 Mar 2004, Terence Parker wrote: I think John's said it all - I have absolutely nothing to add! I'm just posting to second his opinion. Terence On 29 Mar 04, at 3:22 AM, John Baker wrote: -- snip -- I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John ___ 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 ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broken Asterisk
I don't know what I have done, but when I try to start Asterisk I get Ouch Error writing audio data: Broken pipe This scrolls endlessly and I cannot stop the screen except by killing the terminal session. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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] Error installing/compiling cdr_mysql addon
Actually, its MUCH easier to use the unixODBC and cdr_odbc.so modules. That way if you ever change you database to something other than MySQL you wont have to make any major changes to Asterisk. Also, since the ODBC stuff is in the main code instead of the addons you can generally expect it to be better maintained. Michael Shuler, C.E.O. BitWise Systems, Inc. 1301 W. Pioneer Parkway Peoria, IL 61615 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: [EMAIL PROTECTED] Customer Service: (877) 976-0711 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Sunday, March 28, 2004 6:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error installing/compiling cdr_mysql addon When I try to compile the cdr_mysql addon, I get the following error: [EMAIL PROTECTED] asterisk-addons]# make cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I have MySQL installed and have tested it - it is working, I can create databases etc. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error installing/compiling cdr_mysql addon
You need to install the mysql-devel rpm if you use redhat Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Sunday, March 28, 2004 2:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error installing/compiling cdr_mysql addon When I try to compile the cdr_mysql addon, I get the following error: [EMAIL PROTECTED] asterisk-addons]# make cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 I have MySQL installed and have tested it - it is working, I can create databases etc. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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 smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] Broken Asterisk
When you see this message, try to kill mpg123 from another terminal (to stop 'Ouch...') and review the previous errors. Regards, Gus - Original Message - From: Simon Brown [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 28, 2004 10:37 PM Subject: [Asterisk-Users] Broken Asterisk I don't know what I have done, but when I try to start Asterisk I get Ouch Error writing audio data: Broken pipe This scrolls endlessly and I cannot stop the screen except by killing the terminal session. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broken Asterisk
CW_ASN wrote: When you see this message, try to kill mpg123 from another terminal (to stop 'Ouch...') and review the previous errors. And look at the logs too. Asterisk probably was unable to load a module for some reason. Regards, Gus - Original Message - From: Simon Brown [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 28, 2004 10:37 PM Subject: [Asterisk-Users] Broken Asterisk I don't know what I have done, but when I try to start Asterisk I get Ouch Error writing audio data: Broken pipe This scrolls endlessly and I cannot stop the screen except by killing the terminal session. TIA Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as ISDN simulator?
Anyone ever try it? is it possible? I am studying for my CCIE and ISDN simulators are very expensive. Thanks. Will
Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
On Mar 28, 2004, at 7:40 PM, Martin List-Petersen wrote:
; I'm using a shared analog line for testing this, so I'm using the
fax
; autodetection code to yank faxes out of my IVR and into the 'fax'
; pseudo-extension
[outside]
...
exten = fax,1,Goto(fax,2201,1)
I would be interested in how you do fax autodetection.
I don't do anything particularly special, Asterisk just makes it work.
This is using a bog-standard POTS line at home. Here's the relevant
part of my config:
[macro-outsideline]
exten = s,1,LookupCIDName
exten = s,2,SetMusicOnHold(random)
exten = s,3,Dial(${PHONES},13,Ttm)
exten = s,4,Answer
exten = s,5,Goto(outside-ivr,s,1)
[outside-ivr]
; This is the outside IVR
; Playback a We're not home message
; To leave a message for Scott, press 1
; To leave a message for C, press 2
; Otherwise stay on the line.
;
; Also, 3 = main voicemail
; 4 = check voicemail (main)
; 5 = check voicemail
; 6 = DISA (with password)
;
; Check for fax, too
exten = s,1,NoOp
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(2)
exten = s,4,Wait(1)
exten = s,5,Background(laird/ivr-greeting)
exten = t,1,VoiceMail(s2201)
exten = t,2,Hangup
; other stuff goes here, but it's not really important
exten = fax,1,Answer
exten = fax,2,Goto(fax,2201,1)
[outside]
exten = s,1,Macro(outsideline)
exten = fax,1,Goto(fax,2201,1)
95% of this isn't important for faxing, but I included it for context.
The big issue is the IVR stuff and the 'fax' extension. Once we get to
the IVR, asterisk is listening for DTMF tones and apparently also fax
tones. If it hears a fax, then it goes to the 'fax' extension. That's
it.
Scott
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Re: [Asterisk-Users] Codec Voodoo: piece of evidence: probable fix
Hi Ray,
I tried this fix. Did a fresh checkout on 2 servers and established a
SIP-IAX2-SIP call. I still see ocasional jumps of 152 and 168 samples
in the RTP Timestamp. Any other ideas on how to fix this?
Andres
Ray Burkholder wrote:
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
}
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
ms += (now.tv_usec - rtp-txcore.tv_usec) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
return ms;
}
This snippet is from old code. Here is a corrected new snippet with proper
rounding that I think fixes the issue (the two lines are marked [sorry
didn't think to do a diff until afterwards]):
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
rtp-txcore.tv_usec -= rtp-txcore.tv_usec % 2;
}
if (delivery (delivery-tv_sec || delivery-tv_usec)) {
/* Use previous txcore */
=ms = (delivery-tv_sec - rtp-txcore.tv_sec) * 1000;
ms += ((delivery-tv_usec - rtp-txcore.tv_usec) + 500) /
1000;
rtp-txcore.tv_sec = delivery-tv_sec;
rtp-txcore.tv_usec = delivery-tv_usec;
} else {
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
=ms += ((now.tv_usec - rtp-txcore.tv_usec) + 500 ) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
}
return ms;
}
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[Asterisk-Users] Programming an unlocked ADSI Astra 390 phone?
Greetings, I have just purchased several Astra 390 phones ready for asterisk. I have placed a line with adsi=yes in the Zapata.conf file just before channel = 13 I have also added an extension exten = 6199,1,ADSIProg(asterisk.adsi) exten = 6199,2,Hangup in the extensions.conf file. When I try to program the phone I get the following: Asterisk CVS-03/28/04-12:02:10, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-03/28/04-12:02:10 currently running on asterisk (pid = 3328) -- Remote UNIX connection -- Starting simple switch on 'Zap/13-1' -- Executing ADSIProg(Zap/13-1, asterisk.adsi) in new stack -- ADSI Unavailable on CPE. Not bothering to try. -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (local, 6199, 2) exited non-zero on 'Zap/13-1' -- Hungup 'Zap/13-1' asterisk*CLI I am using a Zhone ZPlex 10B channel bank. Why is it telling me that ADSI unavailable on CPE? What do I have to do to get this to work? Also the ADSI documentation is very spotty. Gene Kochanowsky ___ 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] [OT] PoE (Power over Ethernet) for 7940G
Michael Welter wrote: I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM 3CNJPSE power injector. Can I put one of these behind my LAN hub and power all the phones, or do I need one for each phone? From the spec, it looks like PoE tries to discover whether a device is powered over ethernet. Can I just put 48VDC on pins 7-8? Will my Netgear pass this through to the phones? Thanks, If you do this, please have a battery powered video camera ready, I want to see it! No offense intended, honest. A friend of mine insists that if you hook up the power to a floppy drive backwards it creates this nice little poof of blue smoke. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] PoE (Power over Ethernet) for 7940G
Hello, Not sure which pins etc we put it on, but we patched 48V onto our patch panel and did Power over Ethernet that way. Worked well. Regards, Matthew Enger [EMAIL PROTECTED] On Mon, 2004-03-29 at 14:20, Andrew Thompson wrote: Michael Welter wrote: I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM 3CNJPSE power injector. Can I put one of these behind my LAN hub and power all the phones, or do I need one for each phone? From the spec, it looks like PoE tries to discover whether a device is powered over ethernet. Can I just put 48VDC on pins 7-8? Will my Netgear pass this through to the phones? Thanks, If you do this, please have a battery powered video camera ready, I want to see it! No offense intended, honest. A friend of mine insists that if you hook up the power to a floppy drive backwards it creates this nice little poof of blue smoke. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger [EMAIL PROTECTED] Xintegration ___ 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] Error installing/compiling cdr_mysql addon
On 2004 Mar 28, at 19:58, Michael Shuler wrote: Actually, its MUCH easier to use the unixODBC and cdr_odbc.so modules. That way if you ever change you database to something other than MySQL you wont have to make any major changes to Asterisk. Also, since the ODBC stuff is in the main code instead of the addons you can generally expect it to be better maintained. That's nothing short of FUD. cdr_mysql will be maintained for the forseeable future. If you have any concerns about the operation of the module, you're welcome to contact me via email or IRC. -Tilghman ___ 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 as ISDN simulator?
Take a data call in on one BRI and shoot it out on another. Sorry if I was not clear. Would look like this [cisco router with bri][asterisk w 2 bri cards]---[cisco router with bri] I am not to familiar with ISDN so I dunno if I could do this since I know pots has FXO/FXS and you can't go fxo to fxo. Thanks, Will -Original Message- From: Martin List-Petersen [mailto:[EMAIL PROTECTED] Sent: Sunday, March 28, 2004 10:36 PM To: [EMAIL PROTECTED] Cc: william carlson Subject: Re: [Asterisk-Users] Asterisk as ISDN simulator? Citat william carlson [EMAIL PROTECTED]: Anyone ever try it? is it possible? I am studying for my CCIE and ISDN simulators are very expensive. ISDN simulator in what way ? CCIE is far away for me yet, but you can definatly simulate a lot with hfc based cards. /Martin -- BOFH excuse #133: It's not plugged in. ___ 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
AW: [Asterisk-Users] Cisco 7960 SIP Images
I had a completely different experience. The day I decided I wanted to get a contract, I called Cisco, gave them my personal credit card, and three hours later had my CCO access upgraded. I just bought a smartnet for one phone for two years (a whopping $16), there was nothing to it. Nope, same exerience as Johns here. Runaround trying to find a reseller, got a softnet (not smartnet) contract for ~ EUR8 that allows access to CCO; no information available on what else this might include. Waited two weeks for the Service Tokens to arrive by mail only to find out that the online registration site listed on the contracts doesn't know how to handle softnet. Waited another 2 1/2 Weeks for cisco to manually activate the contracts. Cisoc data sheet for the 7940 states: Other Cisco IP Phone 7940G features include: G.711 and G.729a audio compression H.323 compatible and Microsoft NetMeeting compatibility When asked about h.323 compatibility git told that the phone works with cisco call manager, and since that supports h.323, it enables the phone to work with h.323... No comment necessary I think. Bye, Martin ___ 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
