[asterisk-users] Re: codecs/voicemail/DTMF
On 2006-09-20 23:57:09 -0700, Martin Joseph [EMAIL PROTECTED] said: On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said: Hi Eric, I'm confused on where I would put this? I'm also confused on how this would help with external calls (which we want to be g729) vs internal calls to voicemail (which appear to need to be g711)? No, calls to voicemail do not need to be ulaw. You can definitely call voicemail via G729 and use rfc2833 for DTMF. It works depending on your equipment. You are calling using G729 and trying to pass your tones inband, which is impossible due to lack of bandwidth. I think using DTMF=rfc2833 instead of auto is your best bet. Sorry I think that's dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?
The problem is that most people aren't going to be able to answer this question without trying it. Most voip providers (including Teliax) advertise a rate to all New Zealand Mobile service providers, i.e. +64 2, not specifically +64 21xxx. Note, I just tried a +6421 mobile number via Teliax from the U.S. and it worked. So either 1) Teliax can't reliably connect to those numbers, 2) They can connect to some subset of those numbers, or 3) they fixed something since you last checked. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DNS asterisk server?
Hi, I am hating my ISP (comcast) and thinking about switching. One of my options seems pretty good, but doesn't offer a static IP (maybe they will for extra $). Is anyone out there running an asterisk server via dynamic DNS and is this a workable setup? I know my remote ATA's are fine with doing the name lookups, but I wonder if the asterisk server itself will happy behind a nat and a dynamic IP? SIP.conf seems to clearly state that externhost isn't a good way to go and externip is recommended for production environments... This seems like it's a problem for dynamic DNS? Thanks for any experiences and or thoughts on this. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Application of Asterisk Packetization Patch
Pan wrote: I am trying to apply the Asterisk packetization patch found at http://bugs.digium.com/view.php?id=5162 I have several versions of Asterisk, the most recent being 1.2.12.1, but I can not successfully apply the patch. Any suggestions on how to successfully apply this patch to a recent version of Asterisk? This feature has been merged for 1.4, so no patching required I haven't worked on the 1.2.X code in some time, but if you can post the name of the patch you are using and what errors the patch process generates, I can try to help you get it applied. Once the patch is successfully applied I assume I can then set packetization = X on my peers? The latest versions, including the version add to the 1.4 codebase, so the packetization is set on a per codec basis using this format: allow=codec:packetization-ms Examples: allow=ulaw:30 allow=ulaw,g729:40 If :packetization-ms is not added, then the default of 20ms is set. Thanks very much for your time! Cheers, Pan Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and Dial(... line.
exten = 1234567,1,Dial(SIP/[0004f201e443-a) because 0004f201e443-a is
the userid of the phone that you want to send the call to.
As far as I'm concern that isn't acceptable. I would newer make such
configuration. Imagine 1000 extensions and for every one of them you have to
create line like above in extensons.conf.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] multiple zaptel cards
On Thu, Sep 21, 2006 at 11:40:38PM -0500, Jordan Novak wrote: I am in need of an additional x100p in one of my servers. It already has a fully loaded tdm400p in it. I can't figure out how to define the other one in zaptel.conf. Which one do I define first, I am guessing it is dicated by the order the drivers are loaded. I am Wiki'ed out! You are right. This is by the order that they load (or technically: by the order in which their spans register with zaptel). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Picking up a call from queue?
Hi, Is it possible to pick up a call that's in queue and pass it to an agent directly. The use case is that some times some important calls land up in queue which I need to pickup immediatly and pass it on to an agent. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Detection on outbound call
I'm trying to configure my asterisk server to detect fax on an outbound ZAP call. The reason for this is that I have a bunch of interviewers in an outbound callcentre who don't like listening to fax machines and I want to be able to detect fax on the outbound leg before attempting to bridge the call. I have tried using app_nv_faxdetect with the M(faxdetect) option of the Dial command, but I am not sure that this is operating on the right leg of the call. I have tried with /etc/asterisk/Zapata.conf and the faxdetect=... entries, but this doesn't seem to have the desired effect. Does anyone have any suggestions or pointers here? Cheers, Mark. Mark Edwards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P
Are you seeing any IRQ misses Cat /proc/zap/1 and let us know. You might be experiencing some interrupt conflict . M -Original Message- From: David Gagnon [mailto:[EMAIL PROTECTED] Sent: Friday, 22 September 2006 3:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] TDM2400P I had a similar problem and the problem was the hardware echo can who was defect. Try removing the echo can hardware echo can and test the line. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robson Ribeiro Envoyé: 21 septembre 2006 12:28 À: asterisk-users@lists.digium.com Objet: [asterisk-users] TDM2400P Importance: Haute Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycoms IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel = 1-4 channel = 5-8 signalling = fxo_ks channel = 17-20 channel = 21-24 Best Regards, Robson Ribeiro MSN: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows
On 09/20/06 15:06 Dinesh Nair said the following: On 09/19/06 16:59 Steve Langstaff said the following: I wonder whether you are experiencing the following bug (since the SIP INVITE has a multipart SDP body): http://bugs.digium.com/view.php?id=7124nbn=4 thanks for the link, however, on 18th may 2006, kpfleming's note says, This should be fixed in both 1.2 branch and trunk, and i'm using 1.2.12.1 which was just released this week. looking thru the current chan_sip.c code, it does seem like kevin's modified patch has been committed into the branch i'm using, so this isnt the problem. [am cc'ing reply into -dev because a bug report was opened on this at http://bugs.digium.com/view.php?id=8010 with a patch provided] i've managed to track this down to a loop which terminated prematurely in find_sdp() in chan_sip.c. this bug would have prevented proper handling of multipart/mixed content types due to the loop which searches for the end of the block ending prematurely and setting req-sdp_start req-sdp_end. i've provided patches for trunk and 1.2.x in the bug entry, as i think this should also be committed to 1.2.x. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Hi Robert, I use mpg123 version 059r , Fedora Core 5, Shoutcast Server 1.9.7 The streaming bitrate is 56Kb/s mono I have 3 Shoutcast Servers 2 servers over the Internet 128Kb/s / 24Kb/s for public listeners 1 special Shoutcast MOH server in the Asterisk Box || LAN = Asterisk Soutcast Server 56Kb/s mono Streaming source || INTERNET = Public Shoutcast Server 128Kb/s stereo || INTERNET = Public Shoutcast Server 24Kb/s stereo Regards Frédéric Marti = -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Chadwell Sent: jeudi, 21. septembre 2006 19:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Fred, A glimmer of hope! What version of mpg123 do you have running? I am guessing that you control the bitrate on your internal Shoutcast server, is that right? Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frédéric Marti Sent: Thursday, September 21, 2006 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, Yes , I use Asterisk 1.2.10 But , I don't have Warning Flexible rate not heavily tested notices in the Asterisk CLI The Shoutcast server is in the same box as Asterisk, and the stream source is in the same LAN Regards Fred ___ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Chadwell Sent: jeudi, 21. septembre 2006 15:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Frederic, Did this work for you under Asterisk 1.2x? If it did, did you receive Warning Flexible rate not heavily tested notices in the Asterisk CLI? Robert Chadwell 800-330-7704 toll free 813-343-0181 ph 813-413-8195 fx Please feel free to IM me as well AOL Screenname: cmgrobert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk [Submusic] Sent: Wednesday, September 20, 2006 9:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Hi, This config is working for me: _ musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! _ extensions.conf exten = 17,1,Answer exten = 17,2,MusicOnHold(shoutcast) _ Regards Frederic De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robert Chadwell Envoyé : mardi, 19. septembre 2006 14:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Format_MP3, Streaming, File Formats, MOH Format_MP3 appears to play MOH files starting at the beginning of each file, using the .wav file format, making for some repetitive hold music unless you alter the file itself to begin somewhere in the middle. Solution: One stream that all users connect to - giving dynamic hold music (tried and tested in A1.0x using mpg123 with some success, and Icecast or Slimserver or Shoutcast) Format_MP3 doesn't seem to stream, and the wiki is wrong about streamplayer being used to play streams, as it is only used to play raw TCP streams. There are many questions in forums on the web with no answers about how to solve this dilemma, How do you get users connected to a constantly-changing stream of music instead of streams starting from the beginning (regardless of whether Linux counts them as one stream or not where the processor is concerned)? Hopefully, at the end of this thread, I will have enough information to go back to these web-forums and post the answer. To get it started - here is what I have tried that hasn't worked. In most all cases the response is Music on hold started, immediately followed by Music on hold stopped with no sound in any case. ;[classes] ;mode=custom ;application=/usr/bin/streamplayer 194.158.114.67 8000 ;format=ulaw --- Straight From The Music On Hold Wiki ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy -@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls
[asterisk-users] Dual core
Hi list. I have one quick question. Does Asterisk work with dual core processors in version 1.2? Will it work with dual core processors in 1.4? I'm planning to buy new machine for one installation and I have to decide will I buy single or dual core processor. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can you explain why multiple registrationisan important (missing) feature ?
I was afraid that may be the case - The issue I have with that approach is
how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last 4
digits of the DID as the internal extension, I want to use something like
below to handle the bulk of calls:
exten = _,1,Dial(SIP/${EXTEN:4},20)
How can this be accomplished if SIP usernames are mac addresses?, it would
seem to me that sip.conf is the correct place to map an extension to a
device, otherwise I would have an extensions.conf with a manual entry for
each extension making updating it a chore.
Craig
- Original Message -
From: Lacy Moore - Aspendora [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 21, 2006 10:23 AM
Subject: Re: [asterisk-users] Re: Can you explain why multiple
registrationisan important (missing) feature ?
On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote:
[9580]
type=peer
auth=000413242fff:[EMAIL PROTECTED]
It would be
[MAC ADDRESS]
type=peer
...etc..
Or at least, that's how I interpreted what Eric said. I think that's an
excellent approach. THe phones are devices. An extension calls one or
more
devices. Makes a lot more sense than multiple extensions calling multiple
extensions.
Your definition in the sip.conf would be defining devices according to
their
MAC addresses. Your dial plan would call these devices based on
extensions.
exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone
--
Lacy Moore
Aspendora, Inc.
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[asterisk-users] E1 - PCI-Express
Is there any (I prefer one port, but I could also buy two port) E1 PCI-Express card? As far as I can see, all Digim cards are PCI. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forwarding
How might you identify a mobile #? (assuming you refer to cellular phones) Now that phone companies are allowing you to transfer your land line to a mobile, it's no longer practical to use prefix blocking. Where I worked, they just gave up and just restricted forwarding to long distant numbers except by exclusion (for those at the top of the food chain, so to speak) If there is a way to identify, from the number dialed, that the destination is a mobile phone, I'd be interested as well. And curious, why such a preference? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 22 Sep 2006, Paul Hales wrote: I am trying to find a way to stop phones from being forwarded to mobiles - the clients are allowed to forward phones in general, but we want to stop them forwarding calls to mobiles. Is there a SIP header I can check for in the dialplan? I have searched around, but I probably don't quite know what keyword to use in my search... PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] new in 1.4?
Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?
Hi, I'm going to install format_mp3 but I found other two choices, rawplayer and madplay. Anybody knows pros and cons? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iax2 show netstat
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/callaus-3 265 -10-1 -1 0 -1 0
0 40 0 0 00 0
IAX2/2025-4 5 -10-1 -1 0 -1 10
17 92 5 0 10 10
IAX2/callaus-71000 -10-1 -1 0 -1 0
00 0 0 00 0
IAX2/2002-15 4 -10-1 -1 0 -1 12
17 75 3 0 00 11
4 active IAX channels
thanks
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Re: [asterisk-users] Fax Detection on outbound call
On Fri, Sep 22, 2006 at 05:17:03PM +1000, Mark Edwards wrote: I'm trying to configure my asterisk server to detect fax on an outbound ZAP call. The reason for this is that I have a bunch of interviewers in an outbound callcentre who don't like listening to fax machines and I want to be able to detect fax on the outbound leg before attempting to bridge the call. I have tried using app_nv_faxdetect with the M(faxdetect) option of the Dial command, but I am not sure that this is operating on the right leg of the call. I have tried with /etc/asterisk/Zapata.conf and the faxdetect=... entries, but this doesn't seem to have the desired effect. What is the desired effect? What do you get? Can you provide more informamtion on your configuration? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] INVITE re-try interval
Hi Our VoIP provider complains we're sending INVITE retries too quickly. So I think I'm looking for an INVITEequivalent of registertimeout in sip.conf, but there doesn't seem to be one. Any suggestions? David___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sound file length
Raphael Jacquot schrieb: At some point in my dial plan, I need to find out the length of a sound file in seconds (to weed out things that are way too short) the record application doesn't seem to have any facilities to do that. any ideas ? i am wondering ... the voicemail app, does something similar. In voicemail.conf you can specify the minlength of message to it will be processed: voicemail.conf ; Minimum length of a voicemail message in seconds for the message to be ; kept ; The default is no minimum. minmessage=3 Maybe one can have a look at the code of the voicemail-App and tranfer it do the record-App ?? Has this some chance of success ? Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
I was thinking the same thing when reading the press release on sineapps and writing a news article for asteriskguru. I think this covers most of it: - Generic Jitter Buffer - t.38 passthrough - Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks - imap storage for voicemail - whisper paging - Autoconf configuration - menuselect (graphical module select tool similar to the kernel config system) - higher quality prompts (in English, French and Spanish). - watch out they are restructured a little Zoa. Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with wireless client
Sorry, one other equipment query: does anyone know of an ATA with wireless hardware which can act as a *client* to another wireless network? The Linksys units have an integrated wireless access point, but I want something which will work as a client onto an existing wireless network - so you can install ATAs around a building without additional LAN cabling. An ATA with integrated Homeplug (powerline carrier networking) would be another option, but again I can't find such a thing. Any suggestions? Many thanks, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 64 analog phones
Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] freepbx dial plan, add and remove at the same time
Hi, I'm try to setup a dial plan in freepbx to work properly with ENUM lookups. However, the only example I can find that works in the UK is somewhat complex. (http://www.voipuser.org/forum_topic_6651.html) Basically, it has 3 outbound routes (local, national, internation) to strip certain leading digits in a specific order, before a trunk does some more work. I got very close to doing it with a single outbound route (the default, strip the 9, pass the rest) and a single trunk. Where I got stuck was changing 01234567890 into 441234567890. I did see this example: 61+0|NXXX Which to me suggests it will add 61 and strip a leading 0, but either way round it didn't work (even with the correct 10 digits). Can a dial plan infact add and remove numbers at the same time? If so, how? Asterisk 1.2.11, FreePBX 2.1.2. Thanks -- Mike Williams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
quad port T1 card 3 channel banks. Zoa mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to find error codes
Hi,Where can you find error codes tables ?I googled to find that but could find anything.I guess there is something somewhere in source files showing for each error code, a text to display but is there also somewhere suggestions that programmers might leave for systems administators telling them what to check when encountering things like WARNING[3155] chan_zap.c: Call specified, but not found? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iax Netstat Output
can please some one tell me where is what wrong.iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO KpktsIAX2/callaus-3 265 -1 0 -1 -1 0 -1 00 40 0 0 0 0 0IAX2/2025-4 5 -1 0 -1 -1 0 -1 10 17 92 5 0 1 0 10IAX2/callaus-7 1000 -1 0 -1 -1 0 -1 00 0 0 0 0 0 0IAX2/2002-15 4 -1 0 -1 -1 0 -1 12 17 75 3 0 0 0 114 active IAX channelsthanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
Zoa wrote: quad port T1 card 3 channel banks. If expandability isn't a big factor but cost is a dual port E1 card and 2 channel banks. This will get 60 exactly, not 64 tho. [EMAIL PROTECTED] :o) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
2 of xorcom's astribank-32 (http://www.xorcom.com/astribank/features-32.html) On 9/23/06, mike [EMAIL PROTECTED] wrote:Dear listwhich hardware solution would you suggest for connecting 60 analog phones to asterisk ?thank you very much.mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating Asterisk with LDAP Realtime
On Thu, 21 Sep 2006, Nick Couchman wrote: When I try to set the port to 636 in the res_ldap.conf file, I get bind errors (Can't contact server...). I imagine this is an issue with certificates and trust, but I'm not exactly sure where I need to put my CA certificate in order to make the ldap module happy. Probably wherever openssl looks for them. Try /etc/pki/tls/certs/, /etc/ssl/certs/ or /usr/share/ssl/certs/, depending on your distro. You'll also need to symlink the certificate to its hash, check the openssl docs if you haven't done this before. I've tried to use tcpdump to see this data, but tcpdump doesn't grab the full packet, it truncates it at a certain point, so I can't see the data. Try doing your tcpdump with -s 0 - it tells tcpdump to snarf the whole packet Even better, use wireshark (the new name for ethereal). It'll do a very nice job (I tend to find better than tcpdump) at showing you the contents of you ldap queries and responses. I haven't gotten around to playing with direct integration with asterisk and ldap, so I can't help on your other issues. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Two phones, same number
WB == Wes Baehr [EMAIL PROTECTED] writes: WB Use chanisavail to check if one or both phones is busy - if either WB is busy, redirect to voicemail/busy/whatever. Unfortunately chanisavail does not seem to actually ask the phone whether it is busy. When I call it on SIP/somephone, AVAILSTATUS always returns 0. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
On sep/22/2006, mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports. You don't need T1 or E1 extra in the Asterisk machine, only one ethernet card. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk 1.4 going to support realtime ex-girlfriend logic?
Hi all, I was deploying Realtime Extensions when I realised that Realtime Asterisk yet doesn't support ex-girlfriend logic, what made me abandon that implementation! Does Asterisk 1.4 going to support that feature? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote:
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/callaus-3 265 -10-1 -1 0 -1 0
0 40 0 0 00 0
IAX2/2025-4 5 -10-1 -1 0 -1 10
17 92 5 0 10 10
IAX2/callaus-71000 -10-1 -1 0 -1 0
00 0 0 00 0
IAX2/2002-15 4 -10-1 -1 0 -1 12
17 75 3 0 00 11
4 active IAX channels
Could you please tell us why do you believe that there is actually
something wrong?
Or is this a certain Asterisk-competence quiz that I have just failed?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Fax Detection on outbound call
Sure. Agents are logged individually into queues and can therefore work offhook. My application issues an 'originate' via AMI from the queue to the destination number. When the call is answered it is bridged and connects the Agent to the destination party. The desired effect would be that when the application makes a call to the destination party, if it is a fax number, the call can be prevented from sending audio back to the agent. Happy to provide any further information... Cheers Mark. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Friday, 22 September 2006 6:56 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax Detection on outbound call On Fri, Sep 22, 2006 at 05:17:03PM +1000, Mark Edwards wrote: I'm trying to configure my asterisk server to detect fax on an outbound ZAP call. The reason for this is that I have a bunch of interviewers in an outbound callcentre who don't like listening to fax machines and I want to be able to detect fax on the outbound leg before attempting to bridge the call. I have tried using app_nv_faxdetect with the M(faxdetect) option of the Dial command, but I am not sure that this is operating on the right leg of the call. I have tried with /etc/asterisk/Zapata.conf and the faxdetect=... entries, but this doesn't seem to have the desired effect. What is the desired effect? What do you get? Can you provide more informamtion on your configuration? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can the User Know he has voicemail in the Databases.
Hi Users, I'm developing the Voicemail, By flat files I made it, But now I need to do in MySql Databases,In res_mysql.conf and cdr_mysql.conf I given the Database entitesWhile I'm reloading the asterisk server I have arrrived below one message,Can any one tell what this messages means,[cdr_addon_mysql.so] = (MySQL CDR Backend) == Parsing '/etc/asterisk/cdr_mysql.conf': Found -- Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. How to count the mailbox in the CDR or Voicemail _user.Help me And How to know users has a voicemail box in Database..,,-- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS asterisk server?
Martin Joseph wrote: Hi, I am hating my ISP (comcast) and thinking about switching. One of my options seems pretty good, but doesn't offer a static IP (maybe they will for extra $). Is anyone out there running an asterisk server via dynamic DNS and is this a workable setup? I know my remote ATA's are fine with doing the name lookups, but I wonder if the asterisk server itself will happy behind a nat and a dynamic IP? SIP.conf seems to clearly state that externhost isn't a good way to go and externip is recommended for production environments... This seems like it's a problem for dynamic DNS? Thanks for any experiences and or thoughts on this. I was running Trixbox on a dynamic IP behind a NAT and it worked fine. Use something like DynDNS to point your domain name to you IP. Note, SIP is a PITA to configure behind NAT/Firewalls, as you have to open so many ports. IAX is much easier. Regards, Austin. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk and PowerEdge 1950
Make sure you buy it with PCI slots. I overlooked it and the default was PCI-Express. This was for a file server and when I went to put in the SCSI controller, oh, sh*@^$*@$. -- -- Steven http://www.glimasoutheast.org Ryan Amos [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It is almost always better to use a single T1/E1 card when possible to avoid conflicts. A Digium TE2XXP series card sounds like what you would need. The price is usually less than buying 2 single cards. The server itself is fine. It has 2 PCI slots, so if you went with a single card you would be able to expand later should you find the need. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy Sent: Thursday, September 21, 2006 1:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and PowerEdge 1950 hey folks we're planing to install asterisk for a client of ours was just wondering if the Dell's PowerEdge 1950 will take 2 - T1 cards. or if there any recommendations as to which server would be good for our project. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phone references needed
Hi all, Any polycom phone v1.6 IP301 references? I had purchase three new phone and I cant connect them into Asterisk 1.2.11. I do appreciate if some one can point me how and where ? Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phone references needed
(AstATN) wrote: Hi all, Any polycom phone v1.6 IP301 references? I had purchase three new phone and I cant connect them into Asterisk 1.2.11. I do appreciate if some one can point me how and where ? http://www.voip-info.org/wiki-Polycom+Phones Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] freepbx dial plan, add and remove at the same time
Hi Mike,
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
HTH,
Mat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Williams
Sent: 22 September 2006 10:32
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] freepbx dial plan, add and remove
at the same time
Hi,
I'm try to setup a dial plan in freepbx to work properly with
ENUM lookups. However, the only example I can find that works
in the UK is somewhat complex.
(http://www.voipuser.org/forum_topic_6651.html)
Basically, it has 3 outbound routes (local, national,
internation) to strip
certain leading digits in a specific order, before a trunk
does some more
work.
I got very close to doing it with a single outbound route
(the default, strip
the 9, pass the rest) and a single trunk.
Where I got stuck was changing 01234567890 into 441234567890.
I did see this example:
61+0|NXXX
Which to me suggests it will add 61 and strip a leading 0,
but either way
round it didn't work (even with the correct 10 digits).
Can a dial plan infact add and remove numbers at the same
time? If so, how?
Asterisk 1.2.11, FreePBX 2.1.2.
Thanks
--
Mike Williams
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No virus found in this incoming message.
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Date: 21/09/2006
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Re: [asterisk-users] Re: Can you explain why multiple registrationisan important (missing) feature ?
Craig Guy wrote:
I was afraid that may be the case - The issue I have with that approach
is how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last
4 digits of the DID as the internal extension, I want to use something
like below to handle the bulk of calls:
exten = _,1,Dial(SIP/${EXTEN:4},20)
How can this be accomplished if SIP usernames are mac addresses?, it
would seem to me that sip.conf is the correct place to map an extension
to a device, otherwise I would have an extensions.conf with a manual
entry for each extension making updating it a chore.
sip.conf is not the correct place to map extensions to devices.
extensions.conf is is place to do that. You are still stuck in the
mindset that extension == device.
Part of running a PBX, any PBX, is having to map physical devices /
ports to logical extensions.
We have to put in an exten = entry for each extension. It's not that
hard, and only has to be done once for each extension.
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Re: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and Dial(... line.
exten = 1234567,1,Dial(SIP/[0004f201e443-a) because 0004f201e443-a is
the userid of the phone that you want to send the call to.
As far as I'm concern that isn't acceptable. I would newer make such configuration. Imagine 1000 extensions and for every one of them you have to create line like above in extensons.conf.
Correct. If this is an issue then write some scripts to make it easier.
We've not bothered since we only have to do this when the device for
an extension changes or when we have to add/remove an extension. That
does not happen all that often.
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[asterisk-users] Question about SVN-trunk-r43322 and Asterisk Recording Interface
Group Any known problems with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi script? I'm unable to see voicemailsvia the web even though the MWI is flashing and if I look in /var/spool/asterisk/voicemail/default/100/INBOX I do see msg files in that folder. Have not built a system in a while so I must be rusty. Never had problems with install of asterisk and the ARI or vmail.cgi. Thanks again for all the help I have been given over that last few days. Its been a BIG time saver!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ramdonly crash using Realtime Static
I have Asterisk 1.2.12.1 with Realtime Static configuration.Ramdonly when I reload by the Cli command, It crash...I have queues.conf, agents.conf and extensions.conf in the ast_config table (Postgres database) and connect with Asterisk by unixODBC. Any idea? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
There are a couple more that I have run across.- Shared line Apperance support- Users.conf file for simple config of users and devices- follow me application and conf file- Asterisk Builtin mini-HTTP server On 9/22/06, Zoa [EMAIL PROTECTED] wrote: I was thinking the same thing when reading the press release on sineappsand writing a news article for asteriskguru.I think this covers most of it:- Generic Jitter Buffer- t.38 passthrough- Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks- imap storage for voicemail- whisper paging- Autoconf configuration- menuselect (graphical module select tool similar to the kernel configsystem) - higher quality prompts (in English, French and Spanish). - watch outthey are restructured a littleZoa.Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people- Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Detection on outbound call
Here's the dialplan I am using at the moment.
[dialer-test-2]
exten = _X.,1,Set(TIMEOUT(resposnse)=10)
exten = _X.,n,dial(Zap/g1/${EXTEN},60,M(detect-fax^1^2))
exten = _X.,n,noop(back from dial in dialer-test-2)
exten = t,1,noop(timeout)
[macro-detect-fax]
exten = s,1,noop(detecting fax)
exten = s,n,NVFaxDetect(6|nt|2000)
exten = s,n,noop(after NVFaxDetect)
exten = fax,1,noop(got fax)
exten = fax,n,hangup
exten = talk,1,noop(got talk)
exten = talk,n,hangup
At present, NVFaxDetect doesn't appear to pick up fax tone when I dial out
to a fax machine.
Of course this may end up a question re debugging of asterisk dsp.c!
Cheers,
Mark.
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Friday, 22 September 2006 6:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax Detection on outbound call
On Fri, Sep 22, 2006 at 05:17:03PM +1000, Mark Edwards wrote:
I'm trying to configure my asterisk server to detect fax on an outbound
ZAP
call. The reason for this is that I have a bunch of interviewers in an
outbound callcentre who don't like listening to fax machines and I want to
be able to detect fax on the outbound leg before attempting to bridge the
call.
I have tried using app_nv_faxdetect with the M(faxdetect) option of the
Dial
command, but I am not sure that this is operating on the right leg of the
call.
I have tried with /etc/asterisk/Zapata.conf and the faxdetect=... entries,
but this doesn't seem to have the desired effect.
What is the desired effect? What do you get?
Can you provide more informamtion on your configuration?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking about it a bit more, it would be very usefull if the
settings in followme.conf would allow for an entry that points to the
astdb, something like this:
number = family/key
number = family/key
This will allow to use the dialplan to update the number values in
followme.conf, like this:
exten = _*5X.,1,Set(DB(FM${EXTEN:2:1}/${CALLERIDNUM})=${EXTEN:3})
In which case users can call in *51 follwoed the number to follow,
and the DP will add that to FM1, for the second number they would
dial *52 and that would be added to FM2, and so on. In followme.conf
you would then have:
number = FM1/8143
number = FM2/8143
and so on.
And if an entry in the astdb is empty app_followme will ignore it.
Maybe I'm pushing it, but this feature (since it can all be done in
the dialplan without this app) might not have value if one can't use
it because s/he can't have her/his users call in to change the number
values in followme.conf. In which case it means not being able to use
this.
Denis,
There was some discussion around this feature in app_followme in the
IRC chat rooms and it was decided that for at least the 1.4 release of
app_followme, the group wanted to keep it simple and not have this
feature.
We may add this in the future, but I can tell you that I do plan to
realtime enable the application for the 1.6 cycle which probably
gives you more or less the same functionality.
BJ
--
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http://www.btwtech.com/
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Re: [asterisk-users] freepbx dial plan, add and remove at the same time
On Friday 22 September 2006 13:36, Mat Stace wrote:
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
Thanks, but could you explain how that works?
The {EXTEN:1} suggests the first digit is removed, or perhaps more precisely
that's a place holder for the number dialed starting one digit in?
--
Mike Williams
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[asterisk-users] Help with Tieing Outbound calls to Zap Channels
I would like to tie outbound calls from specific extensions to specific zap channels...I have multiple clients in an executive suite and would like to be able to tie lets say extension 1234 to Zap Channels 1 and 2 and extension 5678 to channels 3 and 4 and so on... This so that their caller ID show up properly on outbound calls.. Thanks Kevin J. Steil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CURL
On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 21, 2006 at 08:41:37AM -0700, Elpidio Ramos wrote: Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard. I know this is not a support area so I am only trying to get some clues. I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk? Any clue will be highly appreciated. (I keep trying digium support). Do you actually have the application and function curl? If you do not have it, it may be because libcurl (or its development package) was not availble. You may need a package of the sort of curl-devel or libcurl-dev installed on your system. You do indeed need libcurl-dev in addition to libcurl in order ot have support compiled in. If you don't have libcurl-dev on your system, the configure process will not find the required header files to build the modules. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callback
Hi dudes I read a lot of callback tutorials but I failed to make it work, can any one tell me how to do it in a brief attached with command line, and I will be thanks full . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint status from dialplan?
Can I get the hint status from the dialplan? I am intending to add lit buttons for the parking slots. currently I am using 1.2.11 with 1 parking button and several pickup buttons (speed dials to the parking slots) since 1.4 allows park() to specify a parking slot, I figure that I can remove the park button and just have several buttons for the slots. plan: button assigned to a virtual extension (we will call it 2001) the hint for 2001 will point to parking slot 701. There fore the button will be lit if a call is parked. If the button pressed will call 2001, check the hint status of 701 and either park(701) or ParkedCall(701) depending on the status of the slot. So, Can I get the hint status from the dialplan? Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
BJ, I believe that asteiskdb is before realtime. It does not give the same functionality, since asterisk apps can only update asteriskdb thru the DP, and built in commands. There was some discussion around this feature in app_followme in the IRC chat rooms and it was decided that for at least the 1.4 release of app_followme, the group wanted to keep it simple and not have this feature. We may add this in the future, but I can tell you that I do plan to realtime enable the application for the 1.6 cycle which probably gives you more or less the same functionality. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
John Marvin wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Well, I don't have a solution for the general case (looped playback), but if you are only playing a sine wave, couldn't you use Playtones() instead? It has the ability to play a tone indefinitely until you tell it to stop. John I thought about that. The problem is that I need to be able to play any kind of tone (e.g. warble, etc.). I'm only using a pure tone right now because it's easy to hear the gaps. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] freepbx dial plan, add and remove at the same time
I can have a go at explaining.
I've had a quick dig through my extensions.conf, and I've got it in an
outgoing sipgate dial command.
exten = _0.,1,Dial(SIP/+44${EXTEN:[EMAIL PROTECTED],30,t)
What it does is in the dial command, it sends +44, then the extension which
you dialled, minus the first digit (the leading 0)
Cheers
Mat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Williams
Sent: 22 September 2006 13:49
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] freepbx dial plan,add and
remove at the same time
On Friday 22 September 2006 13:36, Mat Stace wrote:
It's a while since I did this one myself, but I was doing the exact
same thing when using voipbuster (or whichever of it's sisters
services I was using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
Thanks, but could you explain how that works?
The {EXTEN:1} suggests the first digit is removed, or perhaps
more precisely
that's a place holder for the number dialed starting one digit in?
--
Mike Williams
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Re: [asterisk-users] Callback
Khaled Chehab wrote: Hi dudes I read a lot of callback tutorials but I failed to make it work, can any one tell me how to do it in a brief attached with command line, and I will be thanks full .. You will need to give us an example of what you want it to do before that can be done. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] experience with phones locking up uniden and cisco
I am using the latest asterisk 1.2.12 etc... I have uniden UIP-200 phones, Cisco 7960 phones, Cisco 7912 phones, Cisco 7940 phones. It seems like once in a great while (perhaps every other month) All of these phones lock up and have to be rebooted. Are others experiencing this? The UIP-200 has the latest version. The Cisco phones are a couple version back, but its difficult to get the versions from cisco. The phones are not even in use when they lock up. Whats others experience. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Budgetone phones don't show
On Wed, 2006-09-13 at 10:57 +1000, Paul Hales wrote: From memory, it canalmost I used quite a few Grandstreams on a job a while ago, and my memory says that they will do alpha if you are lucky. If not, you get rubbish. My memory also tells me that UPPER CASE worked better than mixed case. It's actually the same as an old-style calculator display. something like: _ | | - |_| The phone *will* happily display any characters that it can with this combination. (Meaning the firmware has provisions for alphanumeric) e.g. u is fine: |_| and L | |_ etc.. but D is trouble for example. So if you *really* wanted to, I guess, you could translate callernames into a combination of displayable characters before passing it on. Cool hack, but it's probably lots cheaper to buy gxp-2000s ;) Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Hello, On 9/21/06, Lee Howard [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Everything is fine when caller use ECM but when ECM isn't in use I often got unusable incoming faxes (much often that it should be). But when I switch back to fax machine that receive faxes perfectly (at least no visible error). The fax machine itself uses ECM, undoubtedly. That's unfortunately not the case. The remote doesn't asks for ECM so that's disabled or missing on that machine. In this situation fax machine is produce better output and I don't know why. Might a better DSP algo? If callers that have quality problems with IAXmodem+HylaFAX don't have problems with the fax machine, then that strongly indicates that something is wrong with your Asterisk setup... corrupting the audio. Usually this is due to resource constriction of the Zap device, zttest scores less than 99.98% is indicative of that situation. I don't find any info that zttest is destructive or not on an active system. I mean that currently active calls are disturbed or not while zttest running. I can't stop system now. I look into zttest source and find that zttest is using /dev/zap/pseudo but I don't know this 'pseudo' channel is related to any 'real' channel or not. Where should be the problem? Is there any solution for improving quality? Any tuning in Asterisk or Hylafax? As I see some people use slinear for iaxmodem and some user use alaw. Which is better? There is no functional difference between using uLaw, alaw, or slinear... except that using slinear reduces the need for conversion... and thus possibly lessens CPU usage very slightly. I see. I leave it on slinear. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [asterisk-biz] UK Male English Voices
Where are yours ? Mark Phillips wrote: Yet another set? I get about 50 downloads a week for mine. Mark On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham [EMAIL PROTECTED] who did all the hard work of recording them, and Jim Credland [EMAIL PROTECTED] for doing all the converting/sound work. Regards Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Will Tatam Email / JID [EMAIL PROTECTED] Web www.netmindz.net PGP Key www.netmindz.net/will/will_tatam.asc Registered Linux user 294695 Linux Counter http://counter.li.org See http://www.jabber.org/ to find out more about the most advanced cross platform, open source enterprise messaging solution ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Male English Voices
Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham [EMAIL PROTECTED] who did all the hard work of recording them, and Jim Credland [EMAIL PROTECTED] for doing all the converting/sound work. Regards Steve The website appears to be down -- Will Tatam Email / JID [EMAIL PROTECTED] Web www.netmindz.net PGP Key www.netmindz.net/will/will_tatam.asc Registered Linux user 294695 Linux Counter http://counter.li.org See http://www.jabber.org/ to find out more about the most advanced cross platform, open source enterprise messaging solution ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote: Hi list! I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the TEI check request message were I was getting errors. Concerned about that I switched to plain vanilla bristuff. Now everything *seems* to be working without errors but I regulary get reports from people trying to call me that they get a signal that the number is not in use or is disconnected. Is anyone else experiencing the same? yep I had the same here with BRIstuffed-0.3.0-PRE-1l it seems to get progressively worse over time. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/22/06, C F [EMAIL PROTECTED] wrote: BJ, I believe that asteiskdb is before realtime. It does not give the same functionality, since asterisk apps can only update asteriskdb thru the DP, and built in commands. There was some discussion around this feature in app_followme in the IRC chat rooms and it was decided that for at least the 1.4 release of app_followme, the group wanted to keep it simple and not have this feature. We may add this in the future, but I can tell you that I do plan to realtime enable the application for the 1.6 cycle which probably gives you more or less the same functionality. Doesn't the new func_realtime allow you to read/write realtime values through DP functions? I believe that it does. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
I have done looping playback and never experienced significant gaps. Earle Clubb wrote: John Marvin wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Well, I don't have a solution for the general case (looped playback), but if you are only playing a sine wave, couldn't you use Playtones() instead? It has the ability to play a tone indefinitely until you tell it to stop. John I thought about that. The problem is that I need to be able to play any kind of tone (e.g. warble, etc.). I'm only using a pure tone right now because it's easy to hear the gaps. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 snom 360 MWI
Just upgraded my * box to 1.2 and don't seem to be able to get MWI working.
Worked with my previous installation. My conf files are the same ( except
for a few 1.2 changes ). I've tried:
In sip.conf
fromuser=Anyname
fromdomain=my * ip
vmexten=7000
in extensions.conf
exten = default,1,VoicemailMain(${CALLERID})
exten = asterisk,1,VoicemailMain(${CALLERID})
exten = unknown,1,VoicemailMain(${CALLERID})
exten = Unknown,1,VoicemailMain(${CALLERID})
exten = _7000,1,VoicemailMain(${CALLERID})
( I've probably added and removed a million other things but its all
becoming a blur now :-) )
Also, when I press the message key to get my voicemail my phone just calls
it's extension so in effect I call myself. In the past on polycom phones I
had to create a macro that checked to see if the number being called was
the same as the calling extension. Do I have to do this for snoms aswell?
I haven't read anything to that fact and believe me, I've read lots. Still
no luck :-(
Any ideas?
Phil.
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Re: [asterisk-users] Two phones, same number
21 Sep 2006 12:15:01 +0200, Benny Amorsen [EMAIL PROTECTED]: I have considered various ways to solve this. One is to make a queue,and only allow one caller in the queue. As far as I can see this won'twork, at least not when I am busy because I did an outgoing call.Another way is to use GROUP() to put the calls in a separate group, and return busy when GROUP_COUNT 0. Unfortunately I am already usingthe GROUP() functionality for something different on those calls --and it seems a call can't be in two GROUP()'s simultaneously. Why not? Use group categories...you can assign two groups to the same channel if they are different categories...Set(GROUP(cat1)=groupname1)Set(GROUP(cat2)=groupname2)if you want to count channels in a group for a specified category, use GROUP_COUNT( [EMAIL PROTECTED]) Hope this helps... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???
See: http://www.voip-info.org/wiki/view/Asterisk+SS7 Jorge Mendoza Jay R. Ashworth wrote: On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote: Hi I need to connect at least 2 (and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done with Digium cards? if not, which box could I use to convert from SS7 to isdn, (could anyone please recomend one of these boxes, and let me know a ballpark price figure) If you're connecting to a carrier's SS7 network, I'm pretty sure you need to be using carrier-lab-approved hardware -- and very probably software -- to do it. Things may have changed since, oh, 5 or 6 years ago when I last paid any close attention to SS7, but last I heard, the ingress ports to that network are not filtered enough for them to let just anyone hook up to them. That said, those links *used* to be V.35 off the terminal equipment; I don't know whether they're using T-spans for them now, but even if they are, I suspect you might need custom *drivers*, not just custom app-level software. But IANASS7E. Cheers, -- jra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forcing Marker bit, because SSRC has changed
Trying again Has anyone an explanation why this error happens? Only hear my echo and not the other side anymore... and the other side can't hear me... Version asterisk 1.2.9 -- Executing Macro(SIP/1001-9c43, stdexten|1010|SIP/1010) in new stack -- Executing Dial(SIP/1001-9c43, SIP/1010|40|o) in new stack -- Called 1010 -- SIP/1010-8035 is ringing -- SIP/1010-8035 answered SIP/1001-9c43 -- Attempting native bridge of SIP/1001-9c43 and SIP/1010-8035 == Forcing Marker bit, because SSRC has changed == Forcing Marker bit, because SSRC has changed cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking about it a bit more, it would be very usefull if the
settings in followme.conf would allow for an entry that points to the
astdb, something like this:
number = family/key
number = family/key
This will allow to use the dialplan to update the number values in
followme.conf, like this:
exten = _*5X.,1,Set(DB(FM${EXTEN:2:1}/${CALLERIDNUM})=${EXTEN:3})
In which case users can call in *51 follwoed the number to follow,
and the DP will add that to FM1, for the second number they would
dial *52 and that would be added to FM2, and so on. In followme.conf
you would then have:
number = FM1/8143
number = FM2/8143
and so on.
And if an entry in the astdb is empty app_followme will ignore it.
Maybe I'm pushing it, but this feature (since it can all be done in
the dialplan without this app) might not have value if one can't use
it because s/he can't have her/his users call in to change the number
values in followme.conf. In which case it means not being able to use
this.
Denis,
There was some discussion around this feature in app_followme in the
IRC chat rooms and it was decided that for at least the 1.4 release of
app_followme, the group wanted to keep it simple and not have this
feature.
We may add this in the future, but I can tell you that I do plan to
realtime enable the application for the 1.6 cycle which probably
gives you more or less the same functionality.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] alternatives to mpg123: format_mp3, rawplayer or madplay?
Giorgio Incantalupo wrote: Hi, I'm going to install format_mp3 but I found other two choices, rawplayer and madplay. Anybody knows pros and cons? TIA Giorgio Incantalupo ___ I used madplay for a month. It crashed once a week, taking asterisk down with it. Not sure what version of asterisk or madplay so results may vary. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike Depends on current and future needs. I like the Quintum Tenor AX. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???
http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channelsOn 22/09/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote: Hi I need to connect at least 2(and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done with Digium cards? if not, which box could I use to convert from SS7 to isdn, (could anyone please recomend one of these boxes, and let me know a ballpark price figure)If you're connecting to a carrier's SS7 network, I'm pretty sure you need to be using carrier-lab-approved hardware -- and very probablysoftware -- to do it.Things may have changed since, oh, 5 or 6 years ago when I last paidany close attention to SS7, but last I heard, the ingress ports to that network are not filtered enough for them to let just anyone hook up tothem.That said, those links *used* to be V.35 off the terminal equipment; Idon't know whether they're using T-spans for them now, but even if they are, I suspect you might need custom *drivers*, not just customapp-level software.But IANASS7E.Cheers,-- jra--Jay R. Ashworth [EMAIL PROTECTED]DesignerBaylink RFC 2100Ashworth AssociatesThe Things I Think'87 e24St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later,they stop having sex with you.-- Jennifer Crusie; _Fast_Women_ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- RegardsRob LithConnection Telecom CC Mobile:+27 (82) 3893332DDI: +27 (21) 6575163Fax: +27 (21) 6575161 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two safe_asterisk processes on the same PBX???
Hi Tzafrir, I prefer to use safe_asterisk even if it is not robustI have never had crashes problems until today. What I want is a little script that sends me a mail when something happens and safe_asterisk seems to do it (I hope). The only problem as I told you is having two safe_asterisk running...but it can be a misconfiguration, so: what are TTY and CONSOLE parameters for? I left them unchanged but maybe that is not the right choice...maybe should I left TTY in blank? I suspect that can cause the two safe_asterisk to run on the same PBX but I'm not sure...what do you think about it? TIA Giorgio Incantalupo Tzafrir Cohen wrote: On Fri, Sep 15, 2006 at 09:14:25AM -0500, Moises Silva wrote: If you want to have a safe asterisk I would recommend using svscan from daemontools package, more wonderfull software of D.J. Bernstein. http://cr.yp.to/daemontools/svscan.html Assumming you really want to live with DJB-style file system. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular). On 9/22/06, Bruce Reeves [EMAIL PROTECTED] wrote: There are a couple more that I have run across.- Shared line Apperance support- Users.conf file for simple config of users and devices - follow me application and conf file- Asterisk Builtin mini-HTTP server On 9/22/06, Zoa [EMAIL PROTECTED] wrote: I was thinking the same thing when reading the press release on sineappsand writing a news article for asteriskguru. I think this covers most of it:- Generic Jitter Buffer- t.38 passthrough- Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks- imap storage for voicemail - whisper paging- Autoconf configuration- menuselect (graphical module select tool similar to the kernel configsystem) - higher quality prompts (in English, French and Spanish). - watch outthey are restructured a little Zoa.Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people- Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialout-trunk vs. dial group
Hi everybody,
Is there any significant difference between using
Macro(dialout-trunk,1,${EXTEN}) and Dial(Zap/g1/${EXTEN})? If so, what
are the differences?
I am not using freePBX, or any variant of it, but want the
functionallity of dialout-trunk. If I define the trunk in zapata.conf,
will using Dial() suffice?
Thanks
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Re: [asterisk-users] Looped message playback
Eric ManxPower Wieling wrote: I have done looping playback and never experienced significant gaps. Can you give me an example of what worked for you? Did the files contain tones or voice? Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk MSN ?
Hi list, Does anyone knows whether Asterisk is able to talk to MSN peers or not, and if yes to what extend? text-only, audio, video? Thanks Yoann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callback
I have an incoming call from pastn number ,the system with deliver it from e1 . So I want to close the line an call him .(callback) Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Friday, September 22, 2006 4:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callback Khaled Chehab wrote: Hi dudes I read a lot of callback tutorials but I failed to make it work, can any one tell me how to do it in a brief attached with command line, and I will be thanks full .. You will need to give us an example of what you want it to do before that can be done. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 snom 360 MWI
[EMAIL PROTECTED] wrote: Just upgraded my * box to 1.2 and don't seem to be able to get MWI working. Worked with my previous installation. My conf files are the same ( except for a few 1.2 changes ). I've tried: In sip.conf fromuser=Anyname fromdomain=my * ip vmexten=7000 Are you missing something like [EMAIL PROTECTED],password in sip.conf? Also, when I press the message key to get my voicemail my phone just calls it's extension so in effect I call myself. In the past on polycom phones I What do you have set in the Snom login preferences, in the mailbox dialog? 7000, or something else? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Display message on voip phone...hint?
Hi all, Can anyone help me... i need to display the cost of a call during a conversation on a sip or iax phone. I see on voip-info that some snom phone support sendtext application, but i don't know how to update the message with the cost on the phone during the conversation. Every suggestion is apreciated. Thx, Bye Bye Ale ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Forcing Marker bit, because SSRC has changed
Can you get an Ethereal trace that captures the RTP streams going to/from Asterisk? If so, you might look for SSRCs changing mid-stream. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Klingler Sent: 22 September 2006 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Forcing Marker bit, because SSRC has changed Trying again Has anyone an explanation why this error happens? Only hear my echo and not the other side anymore... and the other side can't hear me... Version asterisk 1.2.9 -- Executing Macro(SIP/1001-9c43, stdexten|1010|SIP/1010) in new stack -- Executing Dial(SIP/1001-9c43, SIP/1010|40|o) in new stack -- Called 1010 -- SIP/1010-8035 is ringing -- SIP/1010-8035 answered SIP/1001-9c43 -- Attempting native bridge of SIP/1001-9c43 and SIP/1010-8035 == Forcing Marker bit, because SSRC has changed == Forcing Marker bit, because SSRC has changed cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
b'coz I have same setup at other client is working fine no problem.On 9/22/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote: can please some one tell me where is what wrong. iax2 show netstats LOCAL - REMOTE ChannelRTTJitDelLost %DropOOOKpkts JitDelLost %DropOOOKpkts IAX2/callaus-3 265 -10-1-1 0 -10 0 40 0 0 000 IAX2/2025-45 -10-1-1 0 -1 10 17 92 5 0 10 10 IAX2/callaus-71000 -10-1-1 0 -10 00 0 0 000 IAX2/2002-15 4 -10-1-1 0 -1 12 17 75 3 0 00 11 4 active IAX channelsCould you please tell us why do you believe that there is actuallysomething wrong?Or is this a certain Asterisk-competence quiz that I have just failed? --Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755iax:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 64 analog phones
Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports. You don't need T1 or E1 extra in the Asterisk machine, only one ethernet card. thanks very much to everyone for the comments and the suggestions ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS asterisk server?
On Thu, 2006-09-21 at 23:17 -0700, Martin Joseph wrote: Hi, I am hating my ISP (comcast) and thinking about switching. One of my options seems pretty good, but doesn't offer a static IP (maybe they will for extra $). Is anyone out there running an asterisk server via dynamic DNS and is this a workable setup? I know my remote ATA's are fine with doing the name lookups, but I wonder if the asterisk server itself will happy behind a nat and a dynamic IP? SIP.conf seems to clearly state that externhost isn't a good way to go and externip is recommended for production environments... This seems like it's a problem for dynamic DNS? That is the way my server has been working for the past two years. Getting a static IP address in Mexico is very expensive so most of my customers use dynamic DNS to contact their servers. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
Sorry then, I didn't know that, since I don't use realtime. I don't see any reason to introduce another point of failure for a setup that doesn't absolutely need realtime (like a cluster setup). I think my point is still valid, that asteriskdb comes before realtime. Please anybody outthere give some votes on this. Yes I am aware that this might sort of take this post off topic. But still I want to know what everybody esle thinks on this. On 9/22/06, BJ Weschke [EMAIL PROTECTED] wrote: On 9/22/06, C F [EMAIL PROTECTED] wrote: BJ, I believe that asteiskdb is before realtime. It does not give the same functionality, since asterisk apps can only update asteriskdb thru the DP, and built in commands. There was some discussion around this feature in app_followme in the IRC chat rooms and it was decided that for at least the 1.4 release of app_followme, the group wanted to keep it simple and not have this feature. We may add this in the future, but I can tell you that I do plan to realtime enable the application for the 1.6 cycle which probably gives you more or less the same functionality. Doesn't the new func_realtime allow you to read/write realtime values through DP functions? I believe that it does. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
On Fri, 2006-09-22 at 12:35 +0300, Zoa wrote: quad port T1 card 3 channel banks. Zoa Or 2 Astribank-32 () units that connect to the USB port on your server. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looped message playback
Eric ManxPower Wieling wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten = s,1,Playback(tonefile) exten = s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? You have a long gap in your tone file. Eric, You were correct. The file had some header information that should not have been there. I manually stripped of the header so there's only audio data and now the above works fine. Thanks. Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_isdn / chan_sip problems
Hi, I am using Asterisk 1.2 with internal isdn phones connected via a hfcpci card in nt-mode with misdn. Bridging calls from the internal hfcpci via a avmfritz card (also chan_misdn) to the PSTN works flawlessly. However when I use a sip channel to route the outgoing call via voipstunt, it always rings three times and then gives me a busy indication. With my previous configuration, asterisk 1.0.10, zaphfc, chan_capi-cm this was no problem. I thought it was a sip problem and used sip debug but at the moment when the ringing switches to busy no debug messages appear. I also tried a softphone - it works fine with the same config. So I figure that it has something to do with the chan_misdn to chan_sip bridging. Below it the chan_misdn debug trace from the console at the moment when the switch from ringing to busy occurs. Does this tell anybody something that might help with my problem? Do I have a mistake in my misdn configuration? Thanks in advance for any hints. Best regards, Arik console debug trace - hestia*CLI hestia*CLI hestia*CLI P[ 1] *I IND :TIMEOUT oad:23 dad:070712976872 pid:21 state:DIALING P[ 1] -- state: DIALING P[ 1] I SEND:DISCONNECT oad:23 dad:070712976872 pid:21 P[ 1] -- bc_state:BCHAN_ACTIVATED P[ 1] *ec_disable P[ 1] I IND :RELEASE oad: dad: pid:21 state:DIALING P[ 1] hangup_chan P[ 1] - queue_hangup P[ 1] release_chan: bc with l3id: 10042 P[ 1] * RELEASING CHANNEL pid:21 ctx:macro-tsblcr dad:sip oad:23 state: DIALIN G P[ 1] I SEND:RELEASE_COMPLETE oad: dad: pid:21 P[ 1] -- bc_state:BCHAN_CLEANED Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms Reliably Transmitting (no NAT) to 80.239.235.200:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2079ec6b;rport From: Arik sip:[EMAIL PROTECTED];tag=as7a95fade To: sip:[EMAIL PROTECTED] Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines CReliably Transmitting (no NAT) to 80.239.235.200:5060: OPTIONS sip:sip.voipstunt.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport From: asterisk sip:[EMAIL PROTECTED];tag=as439face3 To: sip:sip.voipstunt.com Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX.235.200:5060: Max-Forwards: 70 Date: Fri, 22 Sep 2006 15:50:43 GMTbranch=z9hG4bK2079ec6b;rport Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] ---q: 102 CANCEL hestia*CLI -- SIP read from 80.239.235.200:5060: PTIONS SIP/2.0 200 Ok: 0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport From: asterisk sip:[EMAIL PROTECTED];tag=as439face3 To: sip:sip.voipstunt.com Contact: sip:80.239.235.200:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Supported: User-Agent: (Very nice Sip Registrar Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
On Fri, 22 Sep 2006, Conrad Wood wrote: On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote: Hi list! I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the TEI check request message were I was getting errors. Concerned about that I switched to plain vanilla bristuff. Now everything *seems* to be working without errors but I regulary get reports from people trying to call me that they get a signal that the number is not in use or is disconnected. Is anyone else experiencing the same? yep I had the same here with BRIstuffed-0.3.0-PRE-1l it seems to get progressively worse over time. Thanks for the reply. Well it didn't really get worse, before that I was running bristuff-0.2 that was even worse. Bristuff simply locked up the box every 2-3 days without the florz patch. With the florz patch it was reasonably stable though. Not sure why florz is causing problems on bristuff 0.3 It seems that development on bristuff is stalling a bit, maybe because Asterisk is working on native support of MISDN. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk‑users] Inte grating Asterisk with LDAP Realtime
On Thu 21 Sep 2006 Nick Couchman wrote: When I try to set the port to 636 in the res_ldap.conf file I get bind errors Cant contact server I imagine this is an issue with certificates and trust but Im not exactly sure where I need to put my CA certificate in order to make the ldap module happy. Probably wherever openssl looks for them. Try /etc/pki/tls/certs/ /etc/ssl/certs/ or /usr/share/ssl/certs/ depending on your distro. Youll also need to symlink the certificate to its hash check the openssl docs if you havent done this before. Ive just finished trying this and I still get an error when Asterisk tries to connect. I have a couple other things I need to try I need to try to adjust my CA a little bit but if anyone else has other suggestions for me Id appreciate it. Ive tried to use tcpdump to see this data but tcpdump doesnt grab the full packet it truncates it at a certain point so I cant see the data. Try doing your tcpdump with s 0it tells tcpdump to snarf the whole packet Even better use wireshark the new name for ethereal. Itll do a very nice job I tend to find better than tcpdump at showing you the contents of you ldap queries and responses. I was using ethereal to interpret the data but my servers dont have X on them so its hard to run Ethereal or Wireshark directly on the server. So I use tcpdump to capture to a file then copy to my workstation and use Ethereal to open it. I havent gotten around to playing with direct integration with asterisk and ldap so I cant help on your other issues. NickCouchmanSystemsIntegratorSEAKREngineering,Inc.6221SouthRacineCircleCentennial,CO80111Main:(303)790-8499Fax:(303)790-8720Web:http://www.seakr.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Display message on voip phone...hint?
Just spitballing:
1. Execute a macro in the dial command to spawn an AGI that would return
it's PID to Asterisk and accept the IP address or SIP address of the phone
as an argument. Call the variable, say, ${INCREMENTCOSTPID}
2. The AGI would store call cost variable plus the increment. It would loop
once a minute, and every minute it would execute sipsak with the IP address
of the phone that you returned to the AGI in 1) to display a desktop message
to the phone indicating cost.
3. The h extension would then execute a command like:
exten = h,1,System(kill ${INCREMENTCOSTPID}
Dunno if you can dump a desktop message to a snom while it is on the line -
maybe the snom guys on the list can confirm. Interesting application,
though.
-Original Message-
From: Ale [mailto:[EMAIL PROTECTED]
Sent: Friday, September 22, 2006 9:32 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Display message on voip phone...hint?
Hi all,
Can anyone help me... i need to display the cost of a call during a
conversation on a sip or iax phone.
I see on voip-info that some snom phone support sendtext application,
but i don't know how to update the message with the cost on the phone
during the conversation.
Every suggestion is apreciated.
Thx,
Bye Bye Ale
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[asterisk-users] SNOM 320 - 404 Not Found
I am trying to get a SNOM 320 working with Asierisk. It does register and I can make outbound calls. But it would not take inbound calls. This is what I get; -- Executing Dial(Zap/2-1, SIP/102|20|Tt) in new stack -- Called 102 -- Got SIP response 404 Not Found back from 192.168.1.105 -- SIP/102-cf47 is circuit-busy Here is the outbound call; == Spawn extension (outgoing, 102, 102) exited non-zero on 'SIP/105-5526' -- Executing Dial(SIP/102-fbb7, Zap/g1/9729772921|90) in new stack -- Called g1/NX -- Zap/2-1 answered SIP/102-fbb7 Here is the sip.conf [102] type=friend username=102 secret=102 host=dynamic context=outgoing reinvite=no callwaiting=yes threewaycalling=yes canreinvite=no qualify=300 callerid=102 102 mailbox=102 Here is the sip registration; localhost*CLI sip show peers Name/usernameHost Mask Port Status 102/102 192.168.1.105 (D) 255.255.255.255 5060 OK (41 ms) 101/101 192.168.1.100 (D) 255.255.255.255 5060 OK (40 ms) I would highly appreciate the help to resolve the problem. Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???
On Sep 21, 2006, at 6:15 PM, MF wrote: Hi I need to connect at least 2 (and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done with Digium cards? if not, which box could I use to convert from SS7 to isdn, (could anyone please recomend one of these boxes, and let me know a ballpark price figure) You can check out libss7. I developed it on Digium cards. It is an ss7 implementation in a library that you can use with chan_zap to talk ss7 (like libpri). You also might want to get on the asterisk-ss7 mailing list. There's a lot more talk about this there. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Beta ODBC connection
All my current Asterisk 1.2.12.1 are running on UnixODBC realtime. I just downloaded Asterisk 1.4 beta release this morning and but having problem to compile asterisk with res_odbc on a new server. Have anyone experience this yet and/or hint for me? UnixODBC, UnixODBC-devel and postgresql-odbc are already installed. My system is running on FC4. K ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P and Polycom phones
Dear All, thanks for the help on the TDM2400P. I have resolved the issue. I isolated the problem and ended up finding out it was the Polycom phone that had a problem. Those phones have spectacular quality but they are way too complicated to setup. Also, it's absurd Polycom only supplies you with the latest software if you ask through your reseller!!! What kind of rule is that? Is someone making Polycom phones in China other than Polycom. Well, in any case the phones got confused when selecting the right CODEC to use so I isolated Alaw andDONE. The TDM2400 is fine and works perfectly and so does the phone. One note to people trying to install these phones is that there is a guide on VOIP-Info for the IP500 (more expensive) but not for the Ip301, but they are the same. The other things is that this phone doesn't like to talk to other phones on different networks (like connecting your office and your home with the same phones. Robson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Friday, September 22, 2006 12:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 02:28:40PM -0300, Robson Ribeiro wrote: Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn???t make any difference as the issue doesn???t seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. I apologize; I failed to realize you were non-CONUS; the CNID was odd-looking, and I ignored the call. Feel free to try again. Can you try originating a call out your FXO port from a SIP phone? Is the audio ok when you call FXS to FXS? You need, in general, to use the process of elimination to figure out where your problem *can* be -- even if that entails borrowing hardware. Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4-beta2 Spanish Sounds missing vm-youhaveno?
Hello list! Before I tried the new Asterisk 1.4-beta2 I thought I'd try the 1.4 Spanish sounds on 1.2. When I go to voicemail to get messages it immediately hangs up. Debug shows a missing vm-youhaveno sound file. I took a look at the Asterisk 1.4-beta2 app_voicemail.c and it is still looking to play the vm-youhaveno sound file too. So before I install the 1.4 beta to truly verify the problem is there too, has anyone tried this and have the same problem on the 1.4 beta2 ? earl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback
Khaled Chehab wrote: I have an incoming call from pastn number ,the system with deliver it from e1 . So I want to close the line an call him .(callback) This can be done several ways. The receiving operator can pass the call to a special extension that would either ask for a callback number or read the caller-id from the inbound call and use that number. Once the number is obtained, you create a call file with that number, copy it to the /var/spool/asterisk/outgoing folder and it will call him back and place him in the context that you'd like him to come in from. You could also do this with an extension that automatically answer (Auto attendant) and ask for a password (Authenticate). Info on how to create call files: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Info on how to create IVR: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.confview_comment_id=9484 Info on how to prompt a call for information: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
On 18:22, Fri 22 Sep 06, Remco Barendse wrote: It seems that development on bristuff is stalling a bit, maybe because Asterisk is working on native support of MISDN. Hmm, Will the quad/octobri and gsm cards be supported by MISDN ? I think not. I worked with the cheap HFC-pci bri cards but trashed them all and got some quadbri's. That resolved all my problems. Once again it looks like the golden oneliner is right again: You get what you pay for Just my 2 cents -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
