Re: [Asterisk-Users] unsubscribe
Adam Hart wrote: Try http://lists.digium.com/mailman/listinfo/asterisk-users - Original Message - From: Frank Latini [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 30, 2003 2:18 PM Subject: [Asterisk-Users] unsubscribe Please unsubscribe me from this list people will never learn that they have to unsubscribe themselves !!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nortel PowerTouch 350
It's in line 1 but I also tried line 2 just for kicks but same problem... When you did your tests with different ports and different phones, did you use the same line cord? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html Any idea when these hacks will appear in CVS? We should all hope never. That's why you call it a hack because it works for only one very specific case and would break SIP under Astrisk for most people. It even breaks calls between Asterisk and local SIP phones. Now the trick is to write some code that desides if the trick is to be used or not for each call by comparing the IP address of Asterisk and the called SIP phone. You migh want to experiment with it and report results. Well, I happen to be one of those very specific cases... ;) and looks like will have experiment with it myself. Although I'd hate to re-invent the wheel. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channelbanks for use in europe (Sweden)
At 22:00 29-10-2003 +0100, you wrote: Just remember part of the design of the TE410P is that you can use T1 channel banks (you only get 24 ports) , if you buy these in America they are significantly cheaper than E1 channel banks. Just assign one of the incoming ports on the TE410P to be T1 not E1. unfortunately you loose 6 channels/span and not all t1 channel banks are certified for using in EU Actually for most countries in EU you don't really need certification, especially if you control the other end of the components as well (i.e. a channelbank versus the asterisk box versus the phones you hook up). Should not much of a problem :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Host unspecified ??
Hi Wim, It doesnt show the host (at least) until the phones have registered with asterisk, because you've set the host to dynamic in your config. Either verify if the phones register with asterisk, or set the host to their static IP-adresses. Best regards, Florian At 19:51 29-10-2003 +0100, you wrote: When I start asterisk -vvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop configurations used SIP.CONF [phone1] type=friend host=dynamic defaultip=192.168.10.12 dtmfmode=info mailbox=1000 context=sip callerid=phone1100 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP client
Thanks very much !! I thinks it could be very useful for me Regards Rattana - Original Message - From: Peer Oliver schmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 7:14 PM Subject: Re: [Asterisk-Users] SIP client Christopher Stephens schrieb: Is there SIP client which work with Asterisk and can be embedded in a HTML page ? It may not be *exactly* what you're looking for, but try: http://fwd.pulver.com/callme.php?userid=411 [..] Unfortunately this seem to work with Internet Explorer, only. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Communication between 2 UA
Hello all I setup the Asterisk without Line Card. But UA could not speak each other. Error log was as follow Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Please give us suggestion to communicate each UA. config as follow Asterisk IP address:192.168.0.100 extensions.conf [sip] exten = 1001,1,Dial(SIP/1001,5) exten = 1001,2.Voicemail(u1001) exten = 1001,102.Voicemail(b1001) exten = 1001,103,Hangup exten = 1002,1,Dial(SIP/1002,6) exten = 1002,2.Voicemail(u1002) exten = 1002,102.Voicemail(b1002) exten = 1002,103,Hangup sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 context = sip ;srvlookup = yes ;pedantic = yes Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ;defaultexpirey=120 [1001] type=friend username=1001 secret=secret host=dynamic mailbox=1001 context=sip [1002] type=friend username=1002 secret=secret host=dynamic mailbox=1002 context=sip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Campon feature
Hi Walker,
I've put that up on
http://bugs.digium.com/bug_view_page.php?bug_id=464
Paul
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Campon feature
On Thu, Oct 30, 2003 at 10:28:32AM +1100, Paul Liew wrote:
Hi all,
Having fixed my problems with the call waiting ringing on the GS phones,
I needed to extend that with a campon facility (available on some legacy
systems - sort of callwaiting without phone ringing). I've managed to
implement that by adding/modifying app_queue.c. Basically, when calling the
SIP phone, and if busy, I can camp the call onto that extension, with MOH,
allowing the caller to drop out to voicemail or other priority, if they wish
to. You just need to record an additional voice file as instructions for the
caller in the campon function. Sample of extensions.conf
Paul, this looks great. I'd like to try it.
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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[Asterisk-Users] SIP error: Asked to transmit frame type 64
Hi there,
I'll need some help with this: Trying to establish an IAX2 link between
two servers works in one direction (SIP client with ulaw), but not in the
other (SIP client with GSM). The client used for this is X-Lite behind
NAT while both servers have a public IP (I playback an anouncement before
trying to connect to the second *).
Error on the originating * server:
WARNING[27670]: File chan_sip.c, Line 1148 (sip_write): Asked to transmit
frame type 64, while native formats is 2 (read/write = 2/2)
I really _really_ have no clue why codec 16 bit Signed Linear PCM is n
the game here, to my knowledge that is not supported by X-Lite, and it is
certainly not enabled anyware in the conf files either.
Should I file a bug report, or is this a setup problem on my side?
Philipp
In both sip.conf and iax.conf on both servers I have (with slight
variations):
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
We dial 98616 here:
exten = _9,1,Playback(transfer)
exten = _9,2,Ringing
exten = _9,3,Wait(1)
exten = _9,4,Dial(IAX2/myserv:[EMAIL PROTECTED]/${EXTEN:1})
exten = _9,5,Congestion
exten = _9,105,Playback(tt-monkeysintro)
exten = _9,106,Hangup
my chan_sip.c:
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast-pvt-pvt;
int res = 0;
if (frame-frametype == AST_FRAME_VOICE) {
if (!(frame-subclass ast-nativeformats)) {
-- -- ast_log(LOG_WARNING, Asked to transmit frame type %d, while
native formats is %d (read/write = %d/%d)\n,
frame-subclass, ast-nativeformats, ast-readformat,
ast-
writeformat);
return -1;
}
Related error reports I found:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg12648.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg05602.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg03242.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg01139.html
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Re: [Asterisk-Users] Upcoming Major CVS Changes
Are these changes already done? - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, October 28, 2003 10:32 AM Subject: [Asterisk-Users] Upcoming Major CVS Changes Just as a heads up, soon, I will be merging Thorston Lockhart's new tagged CVS archive over to Digium. This will mean you have to do a *clean* check out of asterisk, zaptel, libpri, etc. For those of you with *localized changes*, please be sure to do: # cvs diff -u ../my-asterisk-changes.diff in the asterisk source directory and keep the resulting file to merge in later on this week with the new Asterisk cvs. We plan to perform the switch Wendsday night or Thursday morning. Again, you only need to do the diff if you're concerned about any changes that you've made locally. To apply the patch once you've checked out the new CVS: # patch -p0 ../my-asterisk-changes.diff As for *why* we're doing this change, and what advantages there will be and how to use them, Thornston will send that information in a separate e-mail. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRAS docs needed...
hi all Where can I find documentation about how to setup ZapRAS? What I want to do (optimally) is to allow for automatic dial-up to external sites, each having an ISDN router. Today we use a small ISDN router for this, but it'd be a lot better, IMHO, to have asterisk do this (functioning as a ISDN router), as we may cancel our BRIs then. Is this possible? And if so, how can I do it? I can't find any docs about ZapRAS at all! roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software FAX
Steve Underwood wrote:
I am taking note of people's messages about soft fax, even if I might
appear to be ignoring them. I am getting V.27ter finished off right
now, to flesh out the facilities in the software. V.27ter is used for
4800bps and 2400bps faxes - not critically important, but useless for
lousy lines. That's seems to be nearly functional now, so I should
soon be back to fixing things.
Most of the crashes seem to be where people have an older version of
libtiff. In each case I've followed up on they have a nice new
libtiff, but still had an old version too. Older versions seem to
cause trouble. I don't intend to find out why, since newer versions
are OK.
An 8 byte TIFF file means it has been opened, and a header written.
The basic TIFF header is always 8 bytes.
I have 8 byte problem only if the file name contains ':' ( if usung
rxfax(/var/spool/fax/${DATETIME} ... )
Regards,
Steve
Brian West wrote:
Good for you... All I can get are 8 byte tiff files.
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the
software
RxFAX/TxFAX up and running without any real problems. I did have
to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include
files to
/usr/local/include
2) build/install spandsp
3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source
tree.
4) move Makefile.patch from oncall to apps/ dir in asterisk
5) patch the makefile
6) edit the makefile and remove all references to steve's home dir
to make
it point to my spandsp source directory.
7) rebuild/install asterisk
8) Create a dir incoming/ in /var/spool/asterisk
9) edit extensions.conf and add the following line to the incoming call
contexts I have set up.
exten =
fax,1,RxFAX(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
10) create a script that emails me the tif files every time they are
received in incoming/ and delete them.
#/bin/sh
cd /var/spool/asterisk/incoming
for X in *.tif
do
if [ -f $X ] ; then
mutt -s FAX from $X -a $X [EMAIL PROTECTED]
/dev/null
rm $X
fi
done
11) Add a cronjob to run my script every 5 minutes.
*/5 * * * * /usr/sbin/mailfax
12) Test and enjoy.
To send a fax all I have to do is
1) Get the .tif file on the server somewhere
2) Put a file sample.call in the /var/spool/asterisk/outgoing/
directory and
it looks like this...
Channel: Zap/3/7989106
Application: txfax
Data: /root/fax.tiff
3) Asterisk will send it or keep trying until it send it as soon as
I :wq
the file in vi.
Pretty simple, I hope this helps someone else.
Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH 43017
Phone: 614-798-9106
FAX: 614-798-9106
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-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]
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[Asterisk-Users] Ringing ....
Hello, I have connected router Cisco 2600 to Asterisk with H323 protocol. Everything is workingfine except... Earlier when I was calling to router, router pick up call and pass it to asterisk IVR system. Then the voice says "Enter your extension" so I enter my extension number, phone is ringing and I can hear ringing tone in my headset. Now after when I update asterisk from CVS (4 days ago) I dial extension,the phone is ringing, but in headset I do not hear ringing tone. What can be wrong?
[Asterisk-Users] Out Of Band DTMF and SIP
I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would really like to know what debug level to use (if any) that would allow me to see that the Phone Event codec packets are being relayed from the Cisco to the GW. Finally, if the GW was unable to convert phone events to DTMF tones, will Asterisk generate the tones on the GW call leg if I configure the SIP phone for out-of-band DTMF and the SIP GW for in-band? Thanks for your help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 pass thru Asterisk
Hello, I have te following setup: IAX client -(iax)- Asterisk -(h323) Cisco AS5300 At the present moment GSM codec is used betwee IAX client and Asterisk. G729 is used between Asterisk and Cisco AS5300. I am thinking that switching from GSM to G729 between IAX client and Asterisk. I know I need G729 licence at the IAX client, but at the Asterisk side can I make Asterisk pass through G729 to Cisco AS5300. This way I do not have to purchace G729 licence for the Asterisk server only for the IAX client. I wonder how this can be done in Asterisk? For example what should I set in the iax.conf or any other .conf file? Reading some of the post in the mailing list, someone mention Asterisk only support passthrough for G723. is that true? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] install problem
Hi, trying to get the make progdocs to work. Got doxygen 1.2.18.3 installed, but during make progdocs I get lots of sh: line 1: dot: command not found And error: problem running dot. Check your installation Need to know how to overcome this, and how to use the documentation afterwards. Another question is where can I get a hold of a decent tutorial for beginners? Thanks.
RE: [Asterisk-Users] Nortel PowerTouch 350
Yes... And I have tried different line cords just rull anything out Does this make sence why this is doing this.. Could it be the phone it self is broke? Geoff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Posted At: Thursday, October 30, 2003 2:00 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Nortel PowerTouch 350 Subject: RE: [Asterisk-Users] Nortel PowerTouch 350 It's in line 1 but I also tried line 2 just for kicks but same problem... When you did your tests with different ports and different phones, did you use the same line cord? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers. It depends on how you did your tests.. If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic.. Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading.. Later.. Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT type router database?
Is anybody aware of a database containing the types of nat implementation in todays soho/consumer routers? I think it would make sense for the community to have this database in order to avoid symmetric nats. If one such thing does not exist how about starting this database? A stunclient for linux can be found at http://sourceforge.net/projects/stun/ I can contribute this information for two routers: W-Linx MB400-X2: coned NAT D-Link DI-604: coned NAT Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 500
Title: Polycom SoundPoint IP 500 The SIP version of the IP500 runs the same firmware, etc as the IP600. The config files are the same. The only difference is that the IP500 has three lines instead of six. I believe that the model number is the same for all IP500 phones, its just the firmware that's different. But, like Matt said, unless you have a copy of the working firmware, I wouldn't try it unless you are willing to potentially render the phone useless in case of an incompatability. -sb -Original Message-From: mattf [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 9:52 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 500 Hello, I only have experience with the IP600 which is SIP only, IP500 is supposedly capable of SIP but you would need to get the firmware from Polycom. I am in the process of trying to sign up for their developer program, but it is a SLOOOW process. I do have the firmware for the IP600 but it is anyone's guess that it would work with the IP500 and I wouldn't want you to ruin your phone trying. The IP600 is a great phone with lots of great features and a good design. Let us know if you get it working. I'll let you know if I get a copy of the IP500 SIP firmware. MATT--- -Original Message-From: Ed Rubright [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 7:52 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Polycom SoundPoint IP 500 Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed
Re: [Asterisk-Users] install problem
Might want to make sure your binaries are in the right place, or at least, where the install script is looking for them - this was my problem. -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Campon feature
Yes, I would like to see the camp feature
become part of the distribution. I know a few people who worked on ROLM
systems who swear there are no replacements just because of some of those features!
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Paul Liew
Sent: Wednesday, October 29, 2003
5:29 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Campon
feature
Hi all,
Having fixed my problems with the
call waiting ringing on the GS phones, I needed to extend that with a campon
facility (available on some legacy systems - sort of callwaiting without phone
ringing). I've managed to implement that by adding/modifying app_queue.c.
Basically, when calling the SIP phone, and if busy, I can camp the call onto
that extension, with MOH, allowing the caller to drop out to voicemail or other
priority,if they wish to.You just need to record an additional
voice file as instructions for the caller in the campon function. Sample of
extensions.conf
[macro-ext]
;
; Standard extension macro:
; ${ARG1} - Technology/Number
;
exten = s,1,Dial(${ARG1},30|tr)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,102,Campon(${ARG1}) ; phone
busy camp the caller on
exten = s,103,Voicemail(b${MACRO_EXTEN}) ; caller decides to leave
voicemail
exten = s,203,Directory(Default) ;
caller decides to call another extension
[extensions]
; our extensions
exten = 2001,1,Macro(ext,SIP/2001)
If there is any interest, I'll post
it up to the bugtracker as a feature ...
Paul
[Asterisk-Users] Info on UK ISDN30e?
Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Gavin Hamill wrote: Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? Cheers, Gavin. AFAIK ISDN30 is the right thing.. BT will provide an RJ45 port for you to connect the Digium card to.. As for the ISDN2's I don't know if its a good idea, especially since the 2 and 4 port BRI cards are very expensive.. Can't help you on the handset question.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On 30/10/03 14:38, Gavin Hamill wrote: Has anyone used ISDN30e in the UK with the Digium E1 cards? Many people. What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. It's more a question of how to implement the backup lines - they're fine for outbound calls, but it's backup for inbound lines that you really want. This is difficult to achieve, but you might be able to get BT to give you a hunt group that hits a pair of ISDN2s after looking through the E1 bank and failing to connect. It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? You will be able to just plug it straight in (standard RJ45 termination). I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Can't answer this one, others? Many people/organizations have successfully deployed it, though. Be aware that it's currently not as easy to configure as many commercial PBXs, but it tends to be cheaper and more flexible. FAX support is also coming soon. :) Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? I doubt you can reuse proprietary handsets. Please provide more details (model/make). -- Alastair Maw MX Telecom www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Linus Magrathea Telecommunications (provider of IAX termination and origination services in the UK!) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip bandwidth usage
This is exactly what I did. I used Xten's GSM driver to call a Zap extension. Readings where 100 Kbits/s. Using uLAW returned 80 Kbits/s !!! I also downloaded Xten pro to test their g729 codec, readings were even worse. That's why I'm so intrigued. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: quinta-feira, 30 de outubro de 2003 10:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip bandwidth usage Paulo Mannheimer wrote: That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers. It depends on how you did your tests.. If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic.. Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading.. Later.. Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
At 14:38 + 30/10/03, Gavin Hamill wrote: The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. You can of course get ISDN30 from other suppliers than BT. Some may try and present as DASS rather than Q931. You want Q931 otherwise you need to get a convertor box. Q931 is the RJ45 version that you just plug in to the line card. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. HTH f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/REGISTER problems!
Hi, I'm trying to get asterisk to work with the Cirpack Softswitch. All I need for now is that asterisk should forward all calls to the Cirpack. My sip.conf files looks like: [general] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 500
Title: Message Hi Matt, Thanks for the reply, it helps alot. I saw your post on 10/20/03 to this list on the review of the Polycom IP 600 review. That was extremely helpful...thanks! Questions: 1) Did you get the call parking issue figured out? From what I could tell from the post it works by pressing #700 (assuming thats your parking extension), but not by pressing the transfer button on the phone? 2) Is there any other feature of the phone that doesn't work for you? 3) You mention in your post that you got the phone for $265 (assuming US $). Is it possible to tell me where? The best I've seen so far is $395. 4) Does the phone work out of the box,or did it require you to flash the phones firmware to get it to work with Asterisk? I like the looks of the Polycom phone and am considering either this a Cisco 7960 or Snom 200 for my SOHO desk set. Thanks, Ed -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattfSent: Wednesday, October 29, 2003 6:52 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 500 Hello, I only have experience with the IP600 which is SIP only, IP500 is supposedly capable of SIP but you would need to get the firmware from Polycom. I am in the process of trying to sign up for their developer program, but it is a SLOOOW process. I do have the firmware for the IP600 but it is anyone's guess that it would work with the IP500 and I wouldn't want you to ruin your phone trying. The IP600 is a great phone with lots of great features and a good design. Let us know if you get it working. I'll let you know if I get a copy of the IP500 SIP firmware. MATT--- -Original Message-From: Ed Rubright [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 7:52 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Polycom SoundPoint IP 500 Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed
[Asterisk-Users] IAX pass url do dialed extension
Hi, I have been working with the Dial application and Gnophone. I would like when the call is placed to the IAX client, an url is passed using the Dial application. I cannot however seem to get the context right to have the url passed onto the GnoPhone answering station. Anyone have a working context? Thanks, James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] critical problem
About every 10th call coming into my x1000p is not getting the audio it should. You can see the messages scrolling on the console as they usually would, playing the thankyou, then and menu messages. internal phones ring, but when answered there is no audio. The caller gets a full volume echo with about 1/2 second latency. At first I thought it might be related to using the aggressive suppressor echo canceller...I recompiled it out, and the problem is still there...doesn't seem to matter if the caller is generating noise when connecting or not. didn't get it to happen when calling from an analog line...seems to happen when calling from a cell phone...I don't see how that would look any different to the x1000p though. perhaps there is a latency difference. I am urgently trying to solve this problem. If I can't solve this problem, it will certainly be the death of my * installation. Has anyone seen this problem before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip bandwidth usage
Paulo Mannheimer wrote: This is exactly what I did. I used Xten's GSM driver to call a Zap extension. Readings where 100 Kbits/s. Using uLAW returned 80 Kbits/s !!! I also downloaded Xten pro to test their g729 codec, readings were even worse. That's why I'm so intrigued. That is odd.. Especially since you got higher bandwidth usage with GSM than you did with G.711.. This is a good site to give you an indication of the bandwidth requirements for various codecs under various conditions.. http://www.packetizer.com/iptel/bandcalc.html Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pass url to dialed extension Stage2
Hi, after hammering out a message, due to several hours of fighting format. I have it resolved. Now, Is there a variable in Extensions that can be used as the incoming callerID from the calling party. i.e. I would like to pass the url, with an attached CallerID string to lookup in our customer database, pulling up the callers record on the agents screen. Thanks, James- James Coberly wrote: Hi, I have been working with the Dial application and Gnophone. I would like when the call is placed to the IAX client, an url is passed using the Dial application. I cannot however seem to get the context right to have the url passed onto the GnoPhone answering station. Anyone have a working context? Thanks, James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP NAT
Should it work to have a multi-homed asterisk server with grandstream phones on the internal network and another grandstream phone on the internet and be able to call between them? I set the bindaddr to the external IP and pointed the internal and external grandstream phones to that address. The signalling works fine to call between phones, but when you pick up the ringing phone you get a reorder tone. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call waiting beep
I am thinking of coding a solution using variables, Cut, and ChanIsAvail. here is what i'm thinking of doing Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ... etc check each phone with ChanIsAvail, and use Cut to remove its representation in the string (if its not avail) then do a dial( variable ) If that doesn't work for some reason, i will try the patch. Thanks for the info. -Sean R. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pass url to dialed extension Stage2
See README.variables in the Asterisk source directory. On Thu, 2003-10-30 at 10:13, James Coberly wrote: Hi, after hammering out a message, due to several hours of fighting format. I have it resolved. Now, Is there a variable in Extensions that can be used as the incoming callerID from the calling party. i.e. I would like to pass the url, with an attached CallerID string to lookup in our customer database, pulling up the callers record on the agents screen. Thanks, James- James Coberly wrote: Hi, I have been working with the Dial application and Gnophone. I would like when the call is placed to the IAX client, an url is passed using the Dial application. I cannot however seem to get the context right to have the url passed onto the GnoPhone answering station. Anyone have a working context? Thanks, James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie with 12sp+
Hi I have problem with Asterisk an 12sp+ phone. Asterisk's skinny implementation doesn't correctly processes 'onhook' event from phone, so voice channel stays opened and no new calls can be received by phone. What i'm doing wrong? :) -- Denis Chapligin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie hardware question
Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a coupleof spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using becauseI will run out of PCI slots if Iuse 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks Jon Hoffman
Re: [Asterisk-Users] Newbie hardware question
the simplest would be to get a t100p card and a 16fxs + 8 fxo channel bank u can find them on ebay quite often, I got mine for 500$ (Carrier access CAC-I with 12fxs and 12fxo) cheers Michael Bielicki On Thursday 30 October 2003 6:00 pm, Just ME wrote: Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got drafted to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a couple of spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using because I will run out of PCI slots if I use 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks Jon Hoffman -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie hardware question
On Thu, 2003-10-30 at 11:00, Just ME wrote: Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got drafted to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a couple of spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using because I will run out of PCI slots if I use 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. The Adit cards are 8 lines at a time. The Adtran unit would let you get 18 extensions and 6 incoming lines on a single T1 interface. Both of these units can be bought on Ebay for relatively inexpensive compared to new prices. Then you will either have to scour the net for the FXO card, or go pay full price for it. Either way, this gets you down to 1 PCI card. If you go the route of a T400P card, adding more service later will be less of a hassle. You could also use it to do your network routing if you decide to go frac T1 for data and some phone service tacked onto the same T1 interface. This could potentially even be a better route as you wouldn't need to find FXO interfaces anymore. You would also get the benefit of using the new software fax setup to get yourself on the way to unified messaging. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie with 12sp+
run a tcpdump -s0 -x tcp port 2000 and send me the results offlist. Jeremy McNamara Denis Chapligin wrote: I have problem with Asterisk an 12sp+ phone. Asterisk's skinny implementation doesn't correctly processes 'onhook' event from phone, so voice channel stays opened and no new calls can be received by phone. What i'm doing wrong? :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP NAT
Dave, Should it work to have a multi-homed asterisk server with grandstream phones on the internal network and another grandstream phone on the internet and be able to call between them? I set the bindaddr to the external IP and pointed the internal and external grandstream phones to that address. The signalling works fine to call between phones, but when you pick up the ringing phone you get a reorder tone. You can probably get it to work. Might read my lengthy rant on nat from the last day or two. If you don't understand the sip protocol in detail, at least recognize that a sip call setup involves: a. sip phone #1 interacts with * on udp 5060 b. after dialing sip phone #2, * will attempt to ask sip phone #1 to contact sip phone #2 directly on udp 5060 (through nat) c. the two sip phones will negotiate some other udp port for the RTP (voice conversation), and the actual port selected is phone-vendor dependent. In your case, the words that you've used suggest the RTP part of that process is being blocked by your nat/firewall box. (That's why you get the reorder tone.) On some sip phones you can set the range of ports to be used for RTP. I'm not a grandstream user, so don't have a clue how that might be done. If it can, then set the range to something like 21000-21010 (or whatever), and set static port forwarding entries in your nat box for the same. May also need nat=yes for the extensions in sip.conf. Another approach is to set canreinvite=no on both phones in sip.conf, which forces the RTP flow to pass through asterisk. (Best check the syntax of that as I'm going from sleep-deprived coffee-lacking memory.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] two things
Hi, I'm having two problems. First I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before? Second thing: I get a WARNING:[1209214400]: File dsp.c, line 1198 (ast_dsp_process): unable to detect process 2 frames All the time. What gives? Please excuse a newbie
Re: [Asterisk-Users] two things
You can only use inband dtmf if you are using the ulaw or alaw codecs. On Thu, 2003-10-30 at 10:46, Shoval Tomer wrote: Hi, I'm having two problems. First I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before? Second thing: I get a WARNING:[1209214400]: File dsp.c, line 1198 (ast_dsp_process): unable to detect process 2 frames All the time. What gives? Please excuse a newbie -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ata-186 vs. TDM400P?
I think I understand the technical side of this, I'm after opions... For a low density Asterisk system (say 3 to 5 extensions) what is the more preferable way to connect analog phones, a small set of Cisco ATA-186 units or a couple Digium TDM400P PCI cards? The criteria are, reliability, sound quality, usability by end users. Yes, I know about hard IP phones but if you need cordless hand sets and other features the IP phone don't have. So what are the practical pros and cons of ata-186 vs. TDM400P? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] two things
Thanks, but no go. I already used these. And it still doesn't work. Anything I can do about the horrible echo in x-lite? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, October 30, 2003 8:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] two things You can only use inband dtmf if you are using the ulaw or alaw codecs. On Thu, 2003-10-30 at 10:46, Shoval Tomer wrote: Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before? Second thing: I get a WARNING:[1209214400]: File dsp.c, line 1198 (ast_dsp_process): unable to detect process 2 frames All the time. What gives? Please excuse a newbie. -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie hardware question
You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. hmm what adtran unit is that the most popular adtran cb's used with * are the ta-750/850 and the slots are provisioned with 4 channels per slot/card total 6 slots per unit, 24 channels total ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Host unspecified ??
Dear, I changed the host to a fixed ip address (host1=192.168.10.12 and host2=192.168.10.13) now the ip address shows up in the 'host' field = ok. Try to call, no succes, nothing happens! What's wrong? Wim - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 30, 2003 9:25 AM Subject: Re: [Asterisk-Users] Host unspecified ?? Hi Wim, It doesnt show the host (at least) until the phones have registered with asterisk, because you've set the host to dynamic in your config. Either verify if the phones register with asterisk, or set the host to their static IP-adresses. Best regards, Florian At 19:51 29-10-2003 +0100, you wrote: When I start asterisk -vvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop configurations used SIP.CONF [phone1] type=friend host=dynamic defaultip=192.168.10.12 dtmfmode=info mailbox=1000 context=sip callerid=phone1100 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie hardware question
One other idea is to go 100% VOIP. Get rid of the incomming analog lines. You can subscribe to a VOIP service that will give you a POTS phone number and route incoming calls to you using SIP. In the office you buy 16 IP hard phones. Now everything is done over Ethernet and you've not got any of those rj-11 jacks. You are also down to zero PCI cards in the computer. You would however need to have a bussiness grade DSL line or fractional T1 installed in the office. I think the above may even be cost effective as you'd be able to skip the expensive channel bank, and T400P. 16 IP phones (Grandstream 101) is only about $1200 --- Steven Critchfield [EMAIL PROTECTED] wrote: On Thu, 2003-10-30 at 11:00, Just ME wrote: Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got drafted to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a couple of spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using because I will run out of PCI slots if I use 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. The Adit cards are 8 lines at a time. The Adtran unit would let you get 18 extensions and 6 incoming lines on a single T1 interface. Both of these units can be bought on Ebay for relatively inexpensive compared to new prices. Then you will either have to scour the net for the FXO card, or go pay full price for it. Either way, this gets you down to 1 PCI card. If you go the route of a T400P card, adding more service later will be less of a hassle. You could also use it to do your network routing if you decide to go frac T1 for data and some phone service tacked onto the same T1 interface. This could potentially even be a better route as you wouldn't need to find FXO interfaces anymore. You would also get the benefit of using the new software fax setup to get yourself on the way to unified messaging. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX gain TX gain
Hi, For me, in order to get the same sound level as for a direct IP/IP call I have the following values: rxgain=10 txgain=15 Unfortunately, with this setting there is a little bit of echo. To get a very small echo but with a lower audio level, the following values work for me: rxgain=0.8 txgain=0.8 By the way... how to interpret those vaules? Thanks, Dan - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 10:37 AM Subject: Re: [Asterisk-Users] RX gain TX gain Lists wrote: I have an X100p cardand it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael Start with 0.5 and see if its too loud or not loud enough and adjust accordingly.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie hardware question
I have 6 750s attached to my pbx server. The 850s have a lot of functionality you don't really need. -sb -Original Message- From: TC [mailto:[EMAIL PROTECTED] Sent: Thursday, October 30, 2003 1:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie hardware question You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. hmm what adtran unit is that the most popular adtran cb's used with * are the ta-750/850 and the slots are provisioned with 4 channels per slot/card total 6 slots per unit, 24 channels total ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
--- Peter Zeltins [EMAIL PROTECTED] wrote: Well, I happen to be one of those very specific cases... ;) and looks like will have experiment with it myself. Although I'd hate to re-invent the wheel. Peter Checking e-mail this morning it looks like we have two independent fixes that both do what has been suggested in this thread. No need for a third except posibly a merge of the two. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SwissVoice MGCP IP10S
I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line = aaln/1 The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,103,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:00:09PM +, Alastair Maw wrote: On 30/10/03 14:38, Gavin Hamill wrote: Has anyone used ISDN30e in the UK with the Digium E1 cards? Many people. That's reassuring to hear :) What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. It's more a question of how to implement the backup lines - they're fine for outbound calls, but it's backup for inbound lines that you really want. Precisely - the outbound stuff we can route via CPS or any Internet SIP provider... We run a call-centre, so it's utterly crucial that the incoming number is always reachable via some method. This is difficult to achieve, but you might be able to get BT to give you a hunt group that hits a pair of ISDN2s after looking through the E1 bank and failing to connect. I think that'll probably be the route we take - I don't imagine management would be very keen on someone routing the call data to us via a network stream - that would require a lot of high-quality, high-expense bandwidth (i.e. UUNet in the UK) It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. I don't quite understand what you mean by this - we want to terminate the ISDN30e ourselves, and have a couple of ISDN2s also there 'just in case'. You will be able to just plug it straight in (standard RJ45 termination). Superb. I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Can't answer this one, others? Many people/organizations have successfully deployed it, though. Ah now that continues in the theme of my question - What people? What organisations? What experiences / issues do they have to tell about the installation? There's the likelihood of a lot of great PR for Asterisk if the relevant parties would only put something in an e-mail :) I mean, I'd love to turn round and be able to say to the bigwigs: 'Hey look, Shell UK have converted their entire nationwide telephony system to Asterisk', but that isn't going to happen - but lots of positive reports from small businesses who have made the switch would (IMO) do Asterisk a power of good. Be aware that it's currently not as easy to configure as many commercial PBXs, but it tends to be cheaper and more flexible. FAX support is also coming soon. :) My previous experience with PBXs was a Siemens HiCom system, and even with the Windows-based config tool, I didn't find it terribly easy to configure. Indeed, I managed to kill the NVRAM on £1500-worth of VoIP card in the process - whoops! or is there the possibilty to re-use proprietary handsets from a previous PBX? I doubt you can reuse proprietary handsets. Please provide more details (model/make). We have an Inter-Tel Axxess unit with 32 extensions at present, and immediate plans for another 16. If I can hook the S0-bus Asterisk and some IP phones as a proof-of-concept, I'd be very happy. Cheers, Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote: I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. I think the point is that we've just about reached capacity on our 30 channels, and won't be in this building for much longer (basically as soon as we can get telephony + data into the new building) so rather than taking another ISDN30, just take a couple of ISDN2s to tide us over in the meantime... isn't eight the minimum no. of channel for a new ISDN30 installation? Q931 is the RJ45 version that you just plug in to the line card. OK thanks for that info, I'll keep it in mind.. I did see DASS2 refered to on BT's ServiceView price list, but didn't know what the alternative was called. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. Yes :) It was only a thought - I think they want to leave the current equipment in the current building in case we need to re-use it at a later date, or suddenly expand past capacity in the new building, etc. Many thanks for your time and comments! Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile problem with older ver. of CVS
While compiling Asterisk from one month ago cvs checkout -D "last month" asterisk I got compiling error: term.c:55: conflicting types for `term_color'include/asterisk/term.h:47: previous declaration of `term_color'term.c:98: conflicting types for `term_prompt'include/asterisk/term.h:49: previous declaration of `term_prompt'make: *** [term.o] Error 1
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Very interesting :) There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told, which we run 4 faxmodems off - I'm not exactly sure /how/ they connect, tbh.. will need to check that out... Perhaps they're just 4 POTS analogue extensions... This would be the ideal testing ground for Asterisk (for me to learn on) since hopefully we could pass the incoming number to the S0-bus, hence Asterisk, hence any IP Phones we buy as a technology demo. The idea of taking a fresh ISDN30 and trying to get everything working from day 1 terrifies me :) We've looked at 'myPBX' from http://www.telappliant.net/site2/mypbx_solution.htm And whilst I like the idea of a pre-configured appliance, I don't know if you get root access, etc. since we will need to write our own applications, etc. As always, I'm open to ideas =) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Q931 is the RJ45 version that you just plug in to the line card. Q931 describes the protocol and not the line presentation. However, you do want to ensure that you ask for Q931 as although DASS/2 is an ISDN protocol, it isnt the same as Euro-ISDN and not supported by Asterisk. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem with older ver. of CVS
On Thu, 2003-10-30 at 20:28, Bartosz Jozwiak wrote: While compiling Asterisk from one month ago cvs checkout -D last month asterisk I got compiling error: term.c:55: conflicting types for `term_color' include/asterisk/term.h:47: previous declaration of `term_color' term.c:98: conflicting types for `term_prompt' include/asterisk/term.h:49: previous declaration of `term_prompt' make: *** [term.o] Error 1 There is an error in either term.c or term.h one has the second variable as a const the other doesn't. If you put the const in it all works fine. Sorry I forgot to post it to bugtrak when I found it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. I don't quite understand what you mean by this - we want to terminate the ISDN30e ourselves, and have a couple of ISDN2s also there 'just in case'. I think the person who replied meant that if you are having the lines as backup in case of failure, you should also be considering failure of the Asterisk equipment and therefore the backup lines should route to a different solution than the ISDN30e / Asterisk one. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem with older ver. of CVS
I just did it. When I call from H323 router and the call is answered I got then segmentation fault. - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: ASTERISK USERS [EMAIL PROTECTED] Sent: Thursday, October 30, 2003 4:41 PM Subject: Re: [Asterisk-Users] Compile problem with older ver. of CVS On Thu, 2003-10-30 at 20:28, Bartosz Jozwiak wrote: While compiling Asterisk from one month ago cvs checkout -D last month asterisk I got compiling error: term.c:55: conflicting types for `term_color' include/asterisk/term.h:47: previous declaration of `term_color' term.c:98: conflicting types for `term_prompt' include/asterisk/term.h:49: previous declaration of `term_prompt' make: *** [term.o] Error 1 There is an error in either term.c or term.h one has the second variable as a const the other doesn't. If you put the const in it all works fine. Sorry I forgot to post it to bugtrak when I found it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Question about MSI 240 Global Station
Is there any one out there using an MSI 240 Global Station with Asterisk? I didn't see it listed on the hardware page but figured I would ask just in case. Thanks, Patrick winmail.dat
Re: [Asterisk-Users] Info on UK ISDN30e?
At 19:24 + 30/10/03, Gavin Hamill wrote: On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote: I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. I think the point is that we've just about reached capacity on our 30 channels, and won't be in this building for much longer (basically as soon as we can get telephony + data into the new building) so rather than taking another ISDN30, just take a couple of ISDN2s to tide us over in the meantime... isn't eight the minimum no. of channel for a new ISDN30 installation? you are asking for an extra bunch of channels, you already have the fibre probably, it may even have the capacity on it just not enabled. Wearing hat with scars on it - don't mess around doing it with ISDN2, the grief qoutient is too high, especially since you won't be there for long and can reuse the kit in the new location. And if you install ISDN2e you will have to take a 12 month contract and have large install costs - you might get away with Business Highway but your cost per channel is probably going to be higher than the ISDN install cost. If you are keeping the current system in place then just explain to account rep that you will be buying new lines from them in new site and keeping on most of the current lines, but what deal will they cut for you. Also remember you have a choice of ISDN providers, even if they are not in your area they can terminate on BT local loop. Q931 is the RJ45 version that you just plug in to the line card. OK thanks for that info, I'll keep it in mind.. I did see DASS2 refered to on BT's ServiceView price list, but didn't know what the alternative was called. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. Yes :) It was only a thought - I think they want to leave the current equipment in the current building in case we need to re-use it at a later date, or suddenly expand past capacity in the new building, etc. Many thanks for your time and comments! nae problem. f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 and G729: Another sad tale
I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. If the answer is No, what is your experience with attempts with these codecs? Are there any workarounds that you have implemented? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Host unspecified ??
Hi Wim, Citeren Wim Venneman [EMAIL PROTECTED]: I changed the host to a fixed ip address (host1=192.168.10.12 and host2=192.168.10.13) now the ip address shows up in the 'host' field = ok. Try to call, no succes, nothing happens! What's wrong? That's a bit difficult to determine without more info. Could you enter the command 'sip debug', try calling with the phones and then copy what the console says ? Feel free to send off-list :-) -- Met vriendelijke groet, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... -- Met vriendelijke groet, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ata-186 vs. TDM400P?
Mixture of 7960's and ATA's for cordless phones... thats what I would do. bkw On Thu, 30 Oct 2003, Chris Albertson wrote: I think I understand the technical side of this, I'm after opions... For a low density Asterisk system (say 3 to 5 extensions) what is the more preferable way to connect analog phones, a small set of Cisco ATA-186 units or a couple Digium TDM400P PCI cards? The criteria are, reliability, sound quality, usability by end users. Yes, I know about hard IP phones but if you need cordless hand sets and other features the IP phone don't have. So what are the practical pros and cons of ata-186 vs. TDM400P? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 07:42:39PM -, Linus Surguy wrote: I think the person who replied meant that if you are having the lines as backup in case of failure, you should also be considering failure of the Asterisk equipment and therefore the backup lines should route to a different solution than the ISDN30e / Asterisk one. Ah of course :) I expect that would be an identical Asterisk installation, and manually switch over the RJ45 cable from box 1 to box 2 if box 1 fails.. Complete duplication of hardware - not very expensive in this case - the biggest expense would probably be the Digium E1 card, and that's a small price to enable such peace of mind! Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX gain TX gain
It's my understand that they are db levels. (And, if I remember my electrical engineering classes from college, a 3db increase effectively doubles the volume.) I hope that helps... Jared Smith On Thu, 2003-10-30 at 11:28, Dan wrote: Hi, For me, in order to get the same sound level as for a direct IP/IP call I have the following values: rxgain=10 txgain=15 Unfortunately, with this setting there is a little bit of echo. To get a very small echo but with a lower audio level, the following values work for me: rxgain=0.8 txgain=0.8 By the way... how to interpret those vaules? Thanks, Dan - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 10:37 AM Subject: Re: [Asterisk-Users] RX gain TX gain Lists wrote: I have an X100p cardand it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael Start with 0.5 and see if its too loud or not loud enough and adjust accordingly.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XTEN-Lite Bad sound!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ariel Batista wrote: | Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I have exactly this same problem as well. It's even worse when running X-Lite under Wine under Linux. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/oXRCuYsUrHkpYtARAsfrAJ4xVad3aMHsDOvarSkeswXml/6U/wCeIdIj i7NmT6VUU/FDvzmay9cyTqY= =tsCQ -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie hardware question
-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Just MESent: Thursday, October 30, 2003 11:00 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Newbie hardware question Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a coupleof spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using becauseI will run out of PCI slots if Iuse 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks Jon Hoffman Jon, Steven just answered this question quite well, so I'll just refer tohim: Andy snip You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. The Adit cards are 8 lines at a time. The Adtran unit would let you get 18 extensions and 6 incoming lines on a single T1 interface. Both of these units can be bought on Ebay for relatively inexpensive compared to new prices. Then you will either have to scour the net for the FXO card, or go pay full price for it. Either way, this gets you down to 1 PCI card. If you go the route of a T400P card, adding more service later will be less of a hassle. You could also use it to do your network routing if you decide to go frac T1 for data and some phone service tacked onto the same T1 interface. This could potentially even be a better route as you wouldn't need to find FXO interfaces anymore. You would also get the benefit of using the new software fax setup to get yourself on the way to unified messaging. -- Steven Critchfield [EMAIL PROTECTED]
Re: [Asterisk-Users] Compile problem with older ver. of CVS
On Thu, 2003-10-30 at 20:53, Bartosz Jozwiak wrote: I just did it. When I call from H323 router and the call is answered I got then segmentation fault. I haven't got any H323 only SIP and analog, I've had no seg faults. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Citeren rnc Info Lists [EMAIL PROTECTED]: I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Were you able to configure the phones through their webinterface ? You could try entering 'mgcp debug' and then power up your phone to see if it registers at all... Yes, web config. of the phone works ok. The IP for the Asterisk server is in the call agent field and port 2427. The following comes on the Asterisk console at powerup. The items between the repeat. MGCP Show endpoints doesn't show anything. Evidently the phone isn't registered but not sure why since there doesn't seem to be a place to associate a userid or password. MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart ** from 192.168.0.5:2427MGCP read: RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1529', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines MGCP read: I RSIP 1529 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Groups in *
Why not just use appqueue? Is that the integrated quesolution that I config in queue.conf? But as I've understood it might be a little tricky to get the users the possibility to log in/out of groups in an easy way (each extension will maybe be the member of up to four groups, and it must be possible to log in/out of each group ...) Then I need the possibility to reroute the call to another group when either all memebers of the group are busy, or when free members in the group don't answer the call for xxx seconds? Is that possible with appqueue? Okay, the solution isn't maybe the best, but it's what the customer wants ... Regards, Lars Fredriksson Hi! Thanks for the tip! Okay, looked a little around AGI and it didn't look to hard doing a script that read which phones that should answer which group from an external textfile, and such file would be quite easy to modify with a CGI-script. And I tried it with a static extensions.conf like below and it seems to work, great! Is there any other considerations or tips about using a solution like this? --SNIP from extensions.conf-- exten = s,1,Answer ; Answer exten = s,2,Dial(Sip/7101Sip/7102,20,m) ; Dial 20 seconds, if busy exten = s,103,Goto(s,3) ; go direct to next group exten = s,3,Dial(Sip/7103Sip/7104,20,m) ; Dial 20 seconds, if busy exten = s,104,Goto(s,4) ; go direct to next group exten = s,4,Dial(Sip/7105Sip/7106,20,m) ; Dial 20 seconds exten = s,5,Goto(s,2); Still no answer, goto ; first group --SNAP-- Regards, Lars - Lars: Anything you want is possible to do with Asterisk... the matter is how much time you want to spend to build that applications... I think that is posible to do that with AGI scripts... Regards, Gus - Original Message - From: Lars Fredriksson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 27, 2003 4:52 PM Subject: [Asterisk-Users] Groups in * Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is busy tha call should stay there for xxx seconds before it goes to another group. But if there are phones free in the group they should ring for xxx seconds before the call goes to another group. And like this it would go on with lots of groups ;-) He also wants queue messages in all groups and the possibility for the phones to log in and out of the different groups (in the morning one phone should be member of three groups, and after lunch log out of those groups and log on to another group ...) I think some kind of web-frontend would be quite kewl, so each employee could log on to a webpage and mark which groups he will answer on (I don't know how * keeps track of such things?) We have tried with PBX's like Panasonic TDA, Ericsson BusinessPhone, Avaya INDeX, Avaya IPOffice and Siemens and none of those can do this ... Thanks for any answer! Best regards Lars Fredriksson, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF x-lite
Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says Premature end of script headers: vmail.cgi Shoval
Re: [Asterisk-Users] Newbie hardware question
You could also look at products like http://sales.netxusa.com/vegastream/vega50.php - Original Message - From: Andy Hester To: [EMAIL PROTECTED] Sent: Thursday, October 30, 2003 3:46 PM Subject: RE: [Asterisk-Users] Newbie hardware question -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Just MESent: Thursday, October 30, 2003 11:00 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Newbie hardware question Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming in and 16 clients (telephones) in our office. I am not worried about the minimum computer requirements because I have a coupleof spare P4 based servers with 512 megs of memory, but I need to know what cards should I be looking at using becauseI will run out of PCI slots if Iuse 4 TDM400P cards (for the clients) and 5 of the X100p (for the lines). Any help or advice would be greatly appreciated. Thanks Jon Hoffman Jon, Steven just answered this question quite well, so I'll just refer tohim: Andy snip You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. The Adit cards are 8 lines at a time. The Adtran unit would let you get 18 extensions and 6 incoming lines on a single T1 interface. Both of these units can be bought on Ebay for relatively inexpensive compared to new prices. Then you will either have to scour the net for the FXO card, or go pay full price for it. Either way, this gets you down to 1 PCI card. If you go the route of a T400P card, adding more service later will be less of a hassle. You could also use it to do your network routing if you decide to go frac T1 for data and some phone service tacked onto the same T1 interface. This could potentially even be a better route as you wouldn't need to find FXO interfaces anymore. You would also get the benefit of using the new software fax setup to get yourself on the way to unified messaging. -- Steven Critchfield [EMAIL PROTECTED]
Re: [Asterisk-Users] Re: call waiting beep
I am thinking of coding a solution using variables, Cut, and ChanIsAvail. here is what i'm thinking of doing Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ... etc check each phone with ChanIsAvail, and use Cut to remove its representation in the string (if its not avail) then do a dial( variable ) If that doesn't work for some reason, i will try the patch. Thanks for the info. -Sean R. I don't think that will work, its been tried before, ChanIsAvail seems to work only for Zap devices. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pass url to dialed extension Stage2
James Coberly wrote: Hi, after hammering out a message, due to several hours of fighting format. I have it resolved. Now, Is there a variable in Extensions that can be used as the incoming callerID from the calling party. i.e. I would like to pass the url, with an attached CallerID string to lookup in our customer database, pulling up the callers record on the agents screen. I don't have this problem, but for archival purposes, would you mind posting how you resolved your problem with examples? Thanks, -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF x-lite
Well, found the answer for the DTMF problem, and guys, the voicemail is G R E A T !!! The answer was use rcf2833 for dtmfmode, not inband as suggested earlier If someone can help me resolve the cgi problem, I'd be forever indebted From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Thursday, October 30, 2003 11:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF x-lite Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says Premature end of script headers: vmail.cgi Shoval
Re: [Asterisk-Users] IAX pass url to dialed extension Stage2
It is shortly explained in README.variables, But for the general
non-readers .
exten =
1112,1,Dial(IAX/[EMAIL PROTECTED]|||http://localhost/bcs/callerid.php?phone=${CALLERIDNUM})
This pops an url to the IAX clients, that queries our customer database
for the client info .
There is also the availablity of ${CALLERID} Name and Number
and ${CALLERIDNAME} Name only.
James
Leif Madsen wrote:
James Coberly wrote:
Hi, after hammering out a message, due to several hours of fighting
format. I have it resolved.
Now, Is there a variable in Extensions that can be used as the
incoming callerID from the calling party.
i.e. I would like to pass the url, with an attached CallerID string
to lookup in our customer database, pulling up the callers record on
the agents screen.
I don't have this problem, but for archival purposes, would you mind
posting how you resolved your problem with examples?
Thanks,
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XTEN-Lite Bad sound!
Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ariel Batista wrote: | Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I have exactly this same problem as well. It's even worse when running X-Lite under Wine under Linux. Has the bad quality started just recently? Has it ever worked nicely for you?? If either of these is yes.. What has changed in your setup? Have you recently upgraded to a newer CVS?? I don't have an answer for you but at least it may stop others falling into the same problem if somthing can be identified as the cause.. later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_oh323
Hello all, can someoen advise what is the exact syntaxt format for the latest OH323 in extensions.conf. we had error when use the chan_oh323. It seems it is a syntaxt error. But we cannot figure out. Please advise if you could. Thanks, below is the detail error info: -- Executing Dial(Zap/4-1, OH323/h323:[EMAIL PROTECTED]|60|r) in new stack WrapH323Connection::WrapH323Connection: WrapH323Connection created. 0:51.230 H323 Cleaner H323Connection ip$localhost/18280 terminated. WrapH323Connection::WrapH323Connection: WrapH323Connection deleted. ERROR[1247605824]: File chan_oh323.c, Line 907 (oh323_call): H323:0: Could not call h323:[EMAIL PROTECTED] -- Couldn't call h323:[EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (pakistan, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' I am using the pwlib 1.5.0, openh323 1.12.0. My extenstions.conf : exten = _X.,1,Dial(OH323/h323:[EMAIL PROTECTED],60,r) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of argnet Sent: Tuesday, June 10, 2003 5:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_oh323 Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten = 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work) Thanx in advance, Victor... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High Availability and Mass Deployment for Asterisk
Senad Jordanovic wrote: Scenario one: One asterisk server, 200+ calls/channels through it. Judging by related posts this scenario will work fine. Scenario two: 1+ calls/channels with one registration URL. I heard that Voyage has 50,000+ clients now. I am talking about that sort of scenario. Mass deployment. What then? 1. Do a lot of switch command to move the calls between servers? 2. Implement a load balancing/high availability solution 3. Your suggestions please Here is my understanding of load balancing: 1. One or more director server are needed which will accept all incoming requests and direct those requests to least busy application server. 2. Two or more application servers running * with shared network file system for all needed directories /var/log/asterisk , /etc/asterisk etc. 3. RAID File Server (RAID 5 preferably) The weakest link would be the director server but if run in a pair that should provide very good reassurance that at least one of them will be running while the faulty one is being replaced. The file server, of course should have its own redundancy put in place. Anyone out there: Is there anything in * operation or structure preventing this sort of setup? Any other suggestions? Ta Senad This is somthing that I have been giving somth thought to as well.. not that I have any need to handle 10 000+ calls but the idea of a clustered PBX is awesome.. Here are some of the issues.. If you are load balancing with a director that spreads the load across multiple servers, the first problem would be sharing the SIP registration information between the two or more servers.. This is so that if UA1 is registered on Server1 and UA2 is registered on Server2.. Then when a call is made from UA1 to UA2, Server1 would know the registration details of UA2 in order to connect the call.. Sure it could be made to try the other servers from within the dialplan but this will be very messy as the number of servers goes up.. Then there is the NAT issue.. If UA1 or UA2 are behind NAT then the NAT table will have an entry for the Server where the UA registered and not the other servers.. So when the call was initiated from one of the other servers the NAT would simply drop the packets.. What is probably needed is one of two things.. a. An SSI(Single System Image) cluster that load balances the processing to multiple nodes but all data in and out are seen to be from a single IP address.. b. A front end router/proxy that handles the IP traffic and SIP information to and from a number of Asterisk nodes behind it that are doing the processing.. Or alternatively a method where by Asterisk is able to be clustered within itself sharing the relevant data and load between the nodes of the cluster and managing the data flow in and out of the server and removing and adding nodes dynamically to the cluster when a node fails or is taken offline and brought back online.. Basically an SSI Asterisk application.. Of course if it did manage this who would need telco's anymore.. :) I guess you could always pop down to the store and order up a 64 way SMP server... that should get at least a couple of thousand concurrent calls going.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323
I had this problem, I believe I fixed it by upgrading openh323, it couldn't parse string for some reason. Unforunately, time has eroded my memory of exact solution/reason. - Original Message - From: G Lin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 9:54 AM Subject: RE: [Asterisk-Users] chan_oh323 Hello all, can someoen advise what is the exact syntaxt format for the latest OH323 in extensions.conf. we had error when use the chan_oh323. It seems it is a syntaxt error. But we cannot figure out. Please advise if you could. Thanks, -- -- below is the detail error info: -- Executing Dial(Zap/4-1, OH323/h323:[EMAIL PROTECTED]|60|r) in new stack WrapH323Connection::WrapH323Connection: WrapH323Connection created. 0:51.230 H323 Cleaner H323Connection ip$localhost/18280 terminated. WrapH323Connection::WrapH323Connection: WrapH323Connection deleted. ERROR[1247605824]: File chan_oh323.c, Line 907 (oh323_call): H323:0: Could not call h323:[EMAIL PROTECTED] -- Couldn't call h323:[EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/4-1, ) in new stack == Spawn extension (pakistan, h, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' I am using the pwlib 1.5.0, openh323 1.12.0. My extenstions.conf : exten = _X.,1,Dial(OH323/h323:[EMAIL PROTECTED],60,r) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of argnet Sent: Tuesday, June 10, 2003 5:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_oh323 Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten = 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work) Thanx in advance, Victor... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Video
Is anyone using Asterisk as the gatekeeper/proxy for videophone calls? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 07:29:17PM +, Gavin Hamill wrote: On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Very interesting :) There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told, which we run 4 faxmodems off - I'm not exactly sure /how/ they connect, tbh.. will need to check that out... Perhaps they're just 4 POTS analogue extensions... S-bus might be ISDN BRI ports, in which case Asterisk can plug in with an AVM Fritz (~110 euro) and chan_capi. This would be the ideal testing ground for Asterisk (for me to learn on) since hopefully we could pass the incoming number to the S0-bus, hence Asterisk, hence any IP Phones we buy as a technology demo. The idea of taking a fresh ISDN30 and trying to get everything working from day 1 terrifies me :) We've looked at 'myPBX' from http://www.telappliant.net/site2/mypbx_solution.htm And whilst I like the idea of a pre-configured appliance, I don't know if you get root access, etc. since we will need to write our own applications, etc. AGI (asterisk gateway interface??) is an application interface for Asterisk, which can use perl, C, php and probably other languages... As always, I'm open to ideas =) A good philosophy. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about IAX/DID's...
Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX DID's, how many calls per DID can an Asterisk box recieve? Is a DID, the same as one line? Or, can multiple people call into each DID at the same time? Regards, Phillip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX gain TX gain
At 10/30/03 12:21 PM, Jared Smith [EMAIL PROTECTED] wrote: It's my understand that they are db levels. (And, if I remember my electrical engineering classes from college, a 3db increase effectively doubles the volume.) As a slight aside on the subject of gain It seems that most people asking about RX/TX gain want to increase their volume. I have the opposite problem: I have a Digium TDM10B FXS card that generates sound far too loud (in the earpiece) with the RX gain set at 0.0, or commented out. That is, routing an analog line = X101P = Asterisk = TDM10B = analog phone is MUCH louder than if I just plug the same phone into the same analog line directly. Some people have suggested that using a negative gain will make it quieter, but I haven't had any luck with this. I *can* make it even louder by increasing the gain -- if I use rxgain = 10 on the TDM10B, for example, it's so loud it sounds like the phone is going to explode -- but using things like rxgain = -3.0 or rxgain = -10.0 doesn't make it any quieter. I can't get it below the rxgain = 0 value. I've been meaning to dig around the source and see what's up, but since it's being discussed... anyone know how to use rxgain to lower the earpiece volume? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN and Asterisk
OK, I've breifly looked at STUN and what it is and can do. First off it is NOT a way to punch UDP through a firewall. STUN offers a method to determine the firewall environment and find out just what is out there. But leaves it to Asterisk to determine what to do. The way it could be used within Asterisk: You would link in the STUN client library from www.vovida.org/ and then when Asterisk first fires up it would call the STUN library to see what kind, if any fire wall is up. It would store this information globally. Later inside chan_sip.c Asterisk could set up the packets correctly with pulic IP address if required. This would be VERY much like the two current patches do except that we would no longer need the new lines in sip.conf as STUN would figure this out for us. The other thing we could do is detect hopeless caes and rather then let the audio fall on the floor we could issue an error message saying something like UDP is 100% blocked no way to make this call and not even attemp it. Bottom line: STUN could save the user much configuration hassel but does noting that a very knowagable person could not figure out and then put into a *.conf file. But most people don't know if their NAT firewall for symetric for restricted cone. STUN can figure this out automatically. Notice that xten X-Lite already does the above. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF x-lite
I've managed to gather that the cgi problem as appears in the httpd error_log is that it can't do setuid. I've searched the web for the last couple of hours and tried almost everything I could find, and I still can't get suexec to work. Can anyone help, please? I know this probably is a newbie question, but the voicemail web interface is a great selling point for the ones upstairs. Thanks a lot for any answer. Shoval From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tom Sent: Friday, October 31, 2003 12:00 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] DTMF x-lite Well, found the answer for the DTMF problem, and guys, the voicemail is G R E A T !!! The answer was use rcf2833 for dtmfmode, not inband as suggested earlier If someone can help me resolve the cgi problem, I'd be forever indebted From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Thursday, October 30, 2003 11:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF x-lite Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says Premature end of script headers: vmail.cgi Shoval
[Asterisk-Users] extension exited non-zero...
Hi, 'g' -- goes on in context if the destination channel hangs up I need the completely opposite of this, something like goes on in context if the calling party hangs up. The situation is as follows, i got a call from outside which is Dial'ed to somewhere else. If the calling party drops the line, the dialplan suddenly ends: == Spawn extension (default, 12999, 4) exited non-zero on 'SIP/dhjk-dbd2' I need to continue the running dialplan to finish the call corrently and clean up things. It seems that there is now way to react to this situation, am i right? Regards, Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? Yes, g729r8 If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. Others seem to have massive issues with chan_h323 and G.729, but i've dealt a dozen or so 5300s of which I haven't had any trouble whatsoever, with nothing other than the code that is currently in the cvs. However, I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. If Asterisk is going to be encoding G.729, yes you will need licenses from Digium. Jeremy McNamara P.S. I'm biased and cannot comment about that other driver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk
Chris, snip OK, I've breifly looked at STUN and what it is and can do. First off it is NOT a way to punch UDP through a firewall. snip Bottom line: STUN could save the user much configuration hassel but does noting that a very knowagable person could not figure out and then put into a *.conf file. But most people don't know if their NAT firewall for symetric for restricted cone. STUN can figure this out automatically. Excellent analysis!!! Can I buy you a beer? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Quoting Jeremy McNamara [EMAIL PROTECTED]: Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? IMHO as for today No, For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300. Calls were dropped from cisco side after two udp packets from cisco sent. I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. Outgoing calls from * to cisco with g729 from digium works fine. But I didnt test it with large volume. regards izo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolute Minimum Installation Packages
why start this with redhat? I'd say it's the worse linux dist to attempt to make a small footprint. Try gentoo. If you wantasterisk with knoppix, then start with that or debian (of which it's based) - Original Message - From: JR Richardson To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 12:31 PM Subject: [Asterisk-Users] Absolute Minimum Installation Packages Im trying to get the total Linux/* installation size as small as possible. Im wondering if anyone has looked at the installed packages list from the Redhat installation [rpm qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. Im sure most of those packages can be eliminated. If the installation can be reduced to below, say 600 Meg, then theres an opportunity to harden * into a KNOPPIX Customization. BTW, has anyone already tried to produce a KNOPPIX * Customization? Thanks in advance. JR
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? IMHO as for today No, For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300. Calls were dropped from cisco side after two udp packets from cisco sent. I've had incoming working with G711, i can't recall if I had it working with G729. I found changing the payload to 20 frames fixed it, although Jeremy tells me I'm crazy and it works without doing that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolute Minimum Installation Packages
JR Richardson wrote: Im trying to get the total Linux/* installation size as small as possible. Im wondering if anyone has looked at the installed packages list from the Redhat installation [rpm qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. Im sure most of those packages can be eliminated. If the installation can be reduced to below, say 600 Meg, then theres an opportunity to harden * into a KNOPPIX Customization. BTW, has anyone already tried to produce a KNOPPIX * Customization? Wierd that I had actually started to just think about this earlier today... :) Unfortunately this is going to be nothing that I can do to help at this point.. I am really quite budgeted for time, and I can barely work on the other things I have somewhat commited to. I'll be so glad when I'm back in school, and hopefully have some more time to work on this kind of stuff. Keep me posted, I have a couple of idea's that this could be useful for (if anything, just what the minimum packages are for a RH install) Thanks! -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Hello, Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? A while ago, I only manage to get g729 call works when terminating in Cisco AS5300 from Asterisk but was unable to terminate call in Asterisk from Cisco AS53000 using g729. If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. not sure about patches, however if you plan to use chan_h323, it is best to get the CORRECT versions of pwlib and openh323 and follow the exact installation instructions. One important thing about these libraries with chan_h323 is DO NOT 'make install' pwlib and openh323 hth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
