Re:[asterisk-users] register suddenly fails
For the time being try putting 212.41.253.181in hostname= line in ur sipconfig and it should work . Also check if you /etc/resolv.conf has correctdns list ( i guess it does bcoz OS canresolve) . Also check /etc/asterisk/dnsmgr.conf . Here's example :[general]enable=yes ; enable creation of managed DNS lookupsrefreshinterval=1200 ; refresh managed DNS lookups every n second - Forwarded message -- From: Norbert Zawodsky norbert at zawodsky.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users at lists.digium.com Date: Thu, 09 Nov 2006 20:28:59 +0100 Subject: [asterisk-users] register suddenly fails Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider inode fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at' timed out, trying again (Attempt #1) Nov 9 20:01:07 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:01:07 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) Nov 9 20:01:27 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at' timed out, trying again (Attempt #2) Nov 9 20:01:28 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:01:28 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds DNS lookup works: root at asterina:~# ping voip.inode.at PING voip.inode.at (212.41.253.181) 56(84) bytes of data. 64 bytes from 212-41-253-181.inhouse-line.inode.at ( 212.41.253.181): icmp_seq=1 ttl=60 time=15.3 ms 64 bytes from 212-41-253-181.inhouse-line.inode.at (212.41.253.181 ): icmp_seq=2 ttl=60 time=15.9 ms --- voip.inode.at ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 15.375/15.669/15.963/0.294 ms Since I am sure that I didn't change anything within the last week, I called inode support. But they said, that they didn't change anything either. Next I tried was a 'SIP RELOAD' which produced following output: asterina*CLI sip reload Nov 9 20:02:48 WARNING[952]: acl.c:244 ast_get_ip_or_srv: Unable to lookup 'h. } ' Nov 9 20:02:48 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:02:48 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) Nov 9 20:03:08 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at ' timed out, trying again (Attempt #1) Nov 9 20:03:08 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:03:08 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) asterina*CLI Now, what makes me wonder ist the first line after the reload which says Unable to lookup 'h. }. Anybody of you got any idea ?? Norbert ___ --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] Voxee lag problems ?
At 08:48 AM 11/9/2006, you wrote: Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem ? So it's not only me! Ira ___ --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] EuroISDN+ and Callers name
I'm running chan_capi on a number of systems in France, France Telecom offer the possibility of having the caller's name, but say we must configure for EuroISDN+. Google doesn't show much and the best I could see was in Dutch. Any Europeans solved this one? Rgds -- Dave Cotton [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] announcing inbound PSTN calls
Hi,This is piece of cake for asterisk, but you need to do your script, or dialplan programing, asterisk has all the functions and applications to do it.But you need to get hands on it :) On 11/10/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere? What is the best approach to doing this? Thanks in advance ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --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] EuroISDN+ and Callers name
Am Freitag, den 10.11.2006, 11:21 +0100 schrieb Dave Cotton: I'm running chan_capi on a number of systems in France, France Telecom offer the possibility of having the caller's name, but say we must configure for EuroISDN+. Google doesn't show much and the best I could see was in Dutch. Any Europeans solved this one? Dave, I do not know about any similar name transmit service in Germany, but obviously there is a possibility to transmit _alpha_numeric caller information via ISDN. I have a Fritz!Box 7050 with internal S0 connected as SIP-client to an asterisk, with a Siemens Gigaset ISDN connected to the FritzBox. In this setup, if I set CALLERID(num) to some string, all the letters and several special cahracters can be transmitted (I wondered why my ISDN display told me the callerid was UNKNOWN, so I set it to an arbitrary string and it worked). This is an out of the box ISDN phone, so it probably is not in any way special. I have no idea though wether Asterisk will understand incoming alpha callerid, or wether mISDN/CAPI do. If you have a reliable online number resolver (the german one is neither reliable nor fast), in your place I would recommend using that. But of course, if you give the FT service a try, please report back about your successes. BR Anselm ___ --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] Station Voip Brazil
OhI'll see you :)On 09/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Felipe Amaral wrote: Hi, There's anyone here who go to Estacao Voip in Brazil??? http://www.estacaovoip.com.br/ I was think to go Anyone here ?? -- Felipe Amaral Vento Livre InternetFelipe,I will be there, and so will Mark :).--Kristian Kielhofner___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Felipe AmaralVento Livre Internet ___ --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] Dropping Connections
Hi! We have an installation with WLAN SIP phones only. Sometimes we have connection drops. What is the best way to debug if we have problems with the WLAN or the SIP devices or the uplink to the IAX Provider. TIA, 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
SV: [asterisk-users] Dropping Connections
Helo My money is on the WLAN part of the equation. We actually dropped WLAN SIP phones altogether, since they worked so poorly. Connection loss, bad audio quality and low coverage range. Just my 5 cents... Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Mike Heininger Sendt: 10. november 2006 12:57 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Dropping Connections Hi! We have an installation with WLAN SIP phones only. Sometimes we have connection drops. What is the best way to debug if we have problems with the WLAN or the SIP devices or the uplink to the IAX Provider. TIA, 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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 09-11-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 09-11-2006 ___ --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] Looking for IP phone / ATA that has builtin VPN support
Dear all, I am looking for ip phone/ ATA that has built in VPN support. can any one suggest me any brand or customize firmware ? thanks Salaque ___ --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] Question on Aastra phones and Astrisk
I've tried other phones and the issue does not happen. I've tried a different IAX provider and it DOES happen... but only if the jitterbuffer is on on the REMOTE side. I am currently working with aastra to try to figure out if this is a phone or asterisk problem. On 11/9/06, shadowym [EMAIL PROTECTED] wrote: That clarifies it! First the stupid questions to eliminate the possibility of anything besides the phones, Have you connected a different make hardphone or softphone and confirmed that works? Have you tried a different IAX/SIP provider? -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 08, 2006 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk It only happens when you go from IAX/SIP -- asterisk box -- aastra phone. Doesn't happen PSTN -- asterisk box -- aastra phone. The aastra people have said they believe it is a codec negotiation issue... but the newest firmware didn't fix it send them packet dumps. On 11/7/06, shadowym [EMAIL PROTECTED] wrote: Running several Aastra 9133i and 480CT phones with v1.4 firmware CentOS 4.4, Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3. Using all default settings I have not seen that problem. I am not exactly sure we are creating those exact same conditions but it sounds like standard extension use to multiple incoming calls correct? That is all we are doing plus some more complicated outgoing stuff. -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 07, 2006 5:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk *bump* Anyone? On 11/6/06, Curt Shaffer [EMAIL PROTECTED] wrote: I wanted to add what we have both seen on traffic captures. You see Caller 1's RTP stream. Call 2 comes in and you see the creation of its RTP stream. After Call 2 is put on hold the RTP stream from Caller 1 disappears without a trace never to return and this is when the one way audio is happening. And I also wanted to add that I am running 1.4.0 firmware for this phone. Thanks again! -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Monday, November 06, 2006 6:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk I'm the friend mentioned here. I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from the PBX to my provider. My issue has a slight twist to it but the same result. For instance his is always where as mine is frequent but not always. After I got to finally see it first hand today, I had to start over from Caller 1 5 times to get it to happen again. Caller 1 calls in and Person A answers. Caller 2 calls in and Person B answers. Person B puts caller 2 on hold and audio drops on Caller 1. So Person A can hear caller 1 but caller 1 cannot hear Person A. This happens more often when Call 1 is on the handset and Call 2 is on the portable or vis a vi, but this is not always the case. It does happen to 1 set only but just less frequent. I have tried carrierinvite=yes and no but this does not change the issue. The phones are behind a router but the external IP of the router is on the same network as the * box. Thanks! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, November 06, 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Aastra phones and Astrisk Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is).We have a connection to and from providers via SIP and IAX.When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me.However, if I park the call, and then pick it up again, the audio is fine. Tonight I tried placing a call on hold using a Sipura/Linksys ATA (that is just hitting 'flash', which basically puts the call on local hold and starts music).The problem did not manifest itself. Has anyone else had this issue? Do you have a fix for it? It is an astrisk issue or an aastra issue? ___ --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:
Re: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer
For what it's worth, Apple's Mail application automatically embeds a tiny QuickTime Player interface in mail messages that contain audio attachments. The result looks like this:http://zachfine.com/blog/images/voicemail_sample_in_apple_mail.gifI'm sure it would not be too difficult to embed a small flash audio player in email messages that has similar functionality. There are some open-source flash media players out there. Here's one that's geared towards playing mp3 streams: http://musicplayer.sourceforge.net/-ZOn 2006-11-10, at 上午7:31, Dean Collins wrote: I know that on my blog I have a flash player which is just html generated from xml feeds.http://deancollinsblog.blogspot.com/ Can a html web page be auto generated from within the Asterisk voicemail module and be sent to an email? What about auto generating a html email with a “player” embeded in the html email? One of the companies I work for (www.tractionplatform.com) do html emails that has a video player in the email so when you open the email in your email clients such as outlook the video streams straight into outlook (email me if you want to see a campaign we ran for Audi – it rocks) The only problem is the video just streams into the html email, there are no player/pause/stop/volume controls in the email. I’ll start a bounty on the wiki with $50 if enough people this is of interest to them and other people can add to it. Cheers, Dean(personally I’m happy to pop an external player with a mp3 because I always have mine running when I’m at my pc but I can see why in companies this might be of interest). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Alex Robar Sent: Tuesday, 7 November 2006 10:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that the message is a VM and not a mail message (so it could display length, date/time received, CID, and provide a "play" button). Just my two cents. Alex On 11/7/06, Dean Collins [EMAIL PROTECTED] wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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 -- Alex Robar [EMAIL PROTECTED]___--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
Re: [asterisk-users] Dropping Connections
Am Freitag, den 10.11.2006, 12:56 +0100 schrieb Mike Heininger: Hi! We have an installation with WLAN SIP phones only. Sometimes we have connection drops. What is the best way to debug if we have problems with the WLAN or the SIP devices or the uplink to the IAX Provider. I'd go with parallel softphones on LAN-connected and/or WLAN-connected PCs and see wether they have the same problem. That could rule out the provider or confirm the WLAN or WLAN phone implementation make the problem. BR Anselm ___ --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] Stable clock with 2.6 and without Digium hardware.
Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable 1000 Hz for some time, but in moments stops/misses interrupts/goes away from 1000 Hz 2.6 with ztdummy USB_UHCI - don't works, needs some tweaking. Somebody knows good patch for it? 2.6 with ztdynamic as primary clock sources - some issues with 2.6 (ztdynamic not ported well to 2.6?) with the mainstream versions, somehow patches solves it. 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. 2.4 with digium card - stable 1000 Hz 2.4 with ztdummy UHCI - stable 1000 Hz 2.4 with ztdynamic clock source - stable 1000 Hz/Depends on network conditions. ___ --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] Queues and Timeouts.
Using Asterisk 1.2.12.1. I have 4 queues running on a server with various handsets logged into them. When a call comes in, asterisk tries forwarding the call to all handsets, including ones that are in use (whereby it gets a BUSY HERE response, which is all what you'd expect after all asterisk doesn't know how many handsets are on each channel). If all the handsets are in use, then asterisk will try calling them every 30 seconds. If I have call-limit=1 in sip.conf, will that mean that asterisk will try to forward the call as soon as a handset becomes free? (including wrapuptime) If this isn't the case is there anything I can set to allow this? Should I set call-limit=1 on the peer definitions as a matter of course anyway? I don't want to just look-and-see since this is running on a production machine and my test machine doesn't have queues installed and is running a completely different version of asterisk. On a slightly different tone, has anyone written a queue viewer that runs as a daemon and serves the pages to the viewer rather than creates a manager login/logout event every few seconds? (If not I'll write one myself, but worth checking first). ___ --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] Stable clock with 2.6 and without Digium hardware.
Can you tell us how you do the testing ? Zoa. Anton Tinchev wrote: Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable 1000 Hz for some time, but in moments stops/misses interrupts/goes away from 1000 Hz 2.6 with ztdummy USB_UHCI - don't works, needs some tweaking. Somebody knows good patch for it? 2.6 with ztdynamic as primary clock sources - some issues with 2.6 (ztdynamic not ported well to 2.6?) with the mainstream versions, somehow patches solves it. 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. 2.4 with digium card - stable 1000 Hz 2.4 with ztdummy UHCI - stable 1000 Hz 2.4 with ztdynamic clock source - stable 1000 Hz/Depends on network conditions. ___ --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: Re: [asterisk-users] Dropping Connections
I'd go with parallel softphones on LAN-connected and/or WLAN-connected PCs and see wether they have the same problem. That could rule out the provider or confirm the WLAN or WLAN phone implementation make the problem. I think this is the next thing we will try next week. TIA, 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] Stable clock with 2.6 and without Digium hardware.
Zoa wrote: Can you tell us how you do the testing ? 3-4 different ways. All gives same results, so test are pretty valid. 1. Interrupt counting inside the PC. 2. TDMoE packet counting on the switch. 3. External TDMoE equipment connected thru extreme network swich. The card of the PC and the device only connected to the switch. The switch filters all packets except TDMoE to de device. Calibrated oscilloscope conected to the interupt leg of the network chip. All coalescing/etc disabling. 4. Diagnostic results from firmware of the device. 5. ToDo test - oscilloscope directly conected to pads inside the PC, but needs mechanical work for each platform/type. 6. ToDo test - some driver relays the clock to simple hardware card in the PC and oscilloscope connected to it. All 4 tests reports same clock difference/clock misses etc. Tested at 3-4 types of hardware - different chipsets/processors. Same results. The are sheduled tests to around 30 more platforms, but pretty sure that the results will be similar. P.S. The device is TDMoE FXS/FXO modular channel bank currently ending development and starting production. ___ --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] VM notification to pager and phone
I looked for a reference to do this for some time to replace the callout
feature in my old AVT voicemail.
I never found one, so I decided to dig in.
Here is my first run. It is in production, so unless I find a problem,
I am done.
Script set to run every 5 min. via cron.
This sets a lock file to prevent 2 scripts from running.
Check for a VM in our Emergency after hours support mailbox.
If found, it sends a numeric page to our rotating pager.
If no one has listened to the mail in 7 minutes, it calls a cell phone.
On this call, it connects directly to a prompt, then VoicemailMain with
the ext. already included.
If no one has listened to the mail in 7 minutes, it calls a second cell
phone.
On this call, it connects directly to a prompt, then VoicemailMain with
the ext. already included.
If no one has listened to the mail in 7 minutes, it calls the rotating
pager again.
This continues to loop until the VM is listened to.
isnotify.sh:
--
LOCKFILE=/tmp/5134outdial.lock
MESSAGEFILE=/var/spool/asterisk/voicemail/default/5134/INBOX/msg.txt
CALLFILE1=/tmp/5134outdial1.call
CALLFILE2=/tmp/5134outdial2.call
CALLFILE3=/tmp/5134outdial3.call
CALLUSER=asterisk
OUTGOING=/var/spool/asterisk/outgoing/
date
# echo lock file check
[ -f $LOCKFILE ] echo $LOCKFILE exists exit 0
touch $LOCKFILE
function recip1 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling IS pager
echo Channel: ZAP/g0/1XXX892 $CALLFILE1
echo MaxRetries: 2 $CALLFILE1
echo RetryTime: 60 $CALLFILE1
echo WaitTime: 30 $CALLFILE1
echo Context: ext-local $CALLFILE1
echo Extension: 5681 $CALLFILE1
echo Priority: 1 $CALLFILE1
echo CallerID: IT VoiceMail XX5682 $CALLFILE1
chown $CALLUSER:$CALLUSER $CALLFILE1
chmod 664 $CALLFILE1
echo move
echo moving $CALLFILE1 to $OUTGOING
mv $CALLFILE1 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip2
}
function recip2 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling BerkHolz
echo Channel: ZAP/g0/1XXX083 $CALLFILE2
echo MaxRetries: 2 $CALLFILE2
echo RetryTime: 60 $CALLFILE2
echo WaitTime: 30 $CALLFILE2
echo Context: ext-local $CALLFILE2
echo Extension: 5682 $CALLFILE2
echo Priority: 1 $CALLFILE2
echo CallerID: IT VoiceMail XX5682 $CALLFILE2
chown $CALLUSER:$CALLUSER $CALLFILE2
chmod 664 $CALLFILE2
echo moving $CALLFILE2 to $OUTGOING
mv $CALLFILE2 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip3
}
function recip3 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling Gibson
echo Channel: ZAP/g0/1XXX061 $CALLFILE3
echo MaxRetries: 2 $CALLFILE3
echo RetryTime: 60 $CALLFILE3
echo WaitTime: 30 $CALLFILE3
echo Context: ext-local $CALLFILE3
echo Extension: 5682 $CALLFILE3
echo Priority: 1 $CALLFILE3
echo CallerID: IT VoiceMail XX5682 $CALLFILE3
chown $CALLUSER:$CALLUSER $CALLFILE3
chmod 664 $CALLFILE3
echo moving $CALLFILE3 to $OUTGOING
mv $CALLFILE3 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip1
}
recip1
rm -f $LOCKFILE
--
Dial Plan:
--
exten = 5681,1,Answer
exten = 5681,n,Wait(3)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(#)
exten = 5681,n,Macro(hangupcall)
exten = 5682,1,Answer
exten = 5682,n,Wait(1)
exten = 5682,n,Macro(user-callerid)
exten = 5682,n,Playback(it-services)
exten = 5682,n,Macro(get-vmcontext,5134)
exten = 5682,n,VoiceMailMain([EMAIL PROTECTED])
exten = 5682,n,Macro(hangupcall)
--
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Board member of
www.glimasoutheast.org
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[asterisk-users] Presence-awareness in Asterisk
Hello all, I am just wondering - how can I implement presence awareness in Asterisk? I know there is the hint feature that might be useful (for someone) but it is not exactly what I am looking for. My idea is some fairly simple application running on user desktop and having just 3-4 buttons like - online - do not disturb - forward to my mobile and possibly also monitoring xscreensaver activity. This application could then communicate with the * server (via AGI or SQL database or something) and amend the dialplan accordingly. Does anyone implemented it somewhere? How can I achieve this? I am happy with just any hint pointing me to the right direction. Thanks, Ondrej ___ --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] Quick Q...
Actually, while I was waiting for an answer, I figured out my problem. If I have any further questions, however, I'll be sure to post. Thanks! Jay Dovid B wrote: Post away. - Original Message - From: Jay Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 09, 2006 6:58 PM Subject: [asterisk-users] Quick Q... Before I make any serious gaffes, is this an acceptable place to post PHPAGI questions as well? I can't seem to find a dedicated mailing list for it. If not, any suggestions? Thanks, Jay ___ --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] Pointers/suggestions?
Hi all,
[I'm new on this list. On other lists, I cringe at this type of query
because they sometimes end up in flame-wars. This is a serious request
because my time frame is about 2 months to identify, select, acquire,
install and setup our next PBX at a new office. I really would like it
to be a VoIP solution.]
Our small real estate firm is experiencing explosive growth lately and
we've outgrown our Nortel ICS 4x12 PBX... I don't like the telco options
and am VERY seriously considering VoIP.
My requirements are:
6-8 fxo
25-50 multi-line hard/soft-phones
mixed Windows/Linux office
Initially, I only need simple telephony features in a pseudo call center
where one/two agents are prime for answering calls from all fxo lines;
but any agent can pick up when things get hectic. Lots of outgoing calls
over these same lines. I also want to eventually integrate fax, e-mail,
etc...
My questions are:
- are there any GPL softphones that can handle 8 lines?
- which hardphone vendors to consider/avoid?
I've been thinking that the live server needs lots of PCI slots; but most
CPU motherboards with 5 slots are discontinued according to the
manufacturer links I follow from Tom's Hardware... Or should I consider
multiple servers?
I have plenty of experience with IP networking (since '87) in all size
networks. My first IP net buildout ('87-'92) was a 300 router network,
and they got bigger when I moved to the US, so this part is the least of
my issues.
Of course, I'm probably forgetting something; so I'm interested in other
gotchas...
Thanks for any pointers,
Pierre
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[asterisk-users] Outgoing problem on PRI
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox
and I am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
Zapata-auto.conf:
callerid=asreceived
signalling=pri_cpe
switchtype=euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
/etc/zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Now I can recieve calls on the pri and everyhting is well but I can't
make calls from the pri, whenever I try I get all circuits are busy message
here is a log from asterisk cli when I try to make a call out using pri
it is a tiny long but trixbox does add many macros and stuff put
I do have suspicions about what can cause the zap channel to get a Hungup
request as it seems from below that is the case :
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")in
new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in newstack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir -UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir - UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheckrecordingcheck|20061110-162404|1163168644.20:
Outbound recording not enabled
recordingcheck|20061110-162404|1163168644.20: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning
0
-- AGI Script recordingcheck completed, returning
0
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/146-
Re: [asterisk-users] asterisk and norstar
Hello Jorge, and thanks for the answers, but:I don't understand what is a blind transfer and a supervised transfer.I mean, in the topology:- pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk
An incoming call from the pstn line is forwarded by the norstar to extension 123 were asterisk is.So asterisk answers the call and play a background message for the caller. But when the user enter the extension number what do we have to do?
I tried with:Hook flash version:exten = _XXX,1,Flash() ;do a hook flash (like pressing FUNCTION in meridian phone)exten = _XXX,2,SendDTMF(*70w${EXTEN},250) ;sends the code for transfer plus the extension
exten = _XXX,3,Hangup()In this version I can transfer the call using the same channel (zap/1) but didn't find a way for voicemail if the call is unanswered or is busy.Also if its unanswered the call is returned to the extension were asterisk is.
Dial version:exten = _XXX,1,Dial(ZAP/1/${EXTEN})It says the channel is busy.I think that with this version I can have a dialstatus for sending to voicemailSo, a couple of questions:What is a blind and a supervised transfer? (cannot find it in the norstar manual)
Do you have and use this topology? if so, how do you do it?Thanks for the help!!(I'm a linux sysadmin and never before worked with telephones system)-- Gustavo BermanSysadminDepto. Informatica
Universidad Nacional del ComahueCentro Regional Universitario Bariloche
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[asterisk-users] Re: Delay between DTMF Down Detected Digit
I have seen this before if the caller is on a cell phone with too high of an
audio delay.
There is a delay for them to hear the end of the prompt, and then a delay for
them sending the digits.
--
--
Steven
http://www.glimasoutheast.org
Jonathan Campbell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call extension 212 and getting connected to the Sales queue which is
option 2 on the IVR. I looked in our log and it seems like there was a
seventeen second delay between the caller pressing the last 2 and when
Asterisk acknowledged it. By that time, Asterisk had decided that 21
wasn't a valid extension and the subsequent 2 dropped the caller into
the Sales queue.
I did my best to search for this issue in the archives and I found one
reference to relaxdtmf, but I wasn't sure if that would address the
issue and I wouldn't want it to cause talkoff.
For reference, we're using a Wildcard TE410P for these incoming calls.
I've included the configuration for the ivr and a scrubbed segment from
the log. If any additional information is needed, please let me know.
Any help is appreciated in advance!
Jon
[ivr-3]
include = ivr-3-custom
include = ext-findmefollow
include = ext-local
include = app-directory
exten = h,1,Hangup
exten = s,1,Set(LOOPCOUNT=0)
exten = s,n,Set(__DIR-CONTEXT=default)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(custom/RM_Daytime)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = 0,1,Goto(ext-queues,300,1)
exten = 1,1,Goto(ext-queues,300,1)
exten = 2,1,Goto(ext-queues,400,1)
exten = 7,1,Goto(ext-queues,700,1)
exten = t,1,Goto(ext-queues,300,1)
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(ivr-3,s,begin)
exten = fax,1,Goto(ext-fax,in_fax,1)
; end of [ivr-3]
Nov 8 11:13:53 VERBOSE[24018] logger.c: -- Accepting call from
'XX' to 's' on channel 0/7, span 1
Nov 8 11:13:53 DEBUG[24018] chan_zap.c: Enabled echo cancellation on
channel 7
...
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131122(131122)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '2'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 262194(262194)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Detected digit '2'
Nov 8 11:13:58 DEBUG[3561] pbx.c: Oooh, got something to jump out with
('2')!
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131121(131121)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '1'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 262193(262193)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Detected digit '1'
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: Got event Event 131122(131122)
on channel 7 (index 0)
Nov 8 11:13:58 DEBUG[3561] chan_zap.c: DTMF Down '2'
Nov 8 11:14:01 VERBOSE[3561] logger.c: -- Invalid extension '21' in
context 'ivr-3' on Zap/7-1
Nov 8 11:14:01 VERBOSE[3561] logger.c: == CDR updated on Zap/7-1
Nov 8 11:14:01 VERBOSE[3561] logger.c: -- Executing
Playback(Zap/7-1, invalid) in new stack
Nov 8 11:14:01 DEBUG[3561] channel.c: Scheduling timer at 160 sample
intervals
Nov 8 11:14:01 DEBUG[24018] chan_zap.c: Echo cancellation already on
...
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Exception on 23, channel 7
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Got event Event 262194(262194)
on channel 7 (index 0)
Nov 8 11:14:15 DEBUG[3561] chan_zap.c: Detected digit '2'
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Re: [asterisk-users] Presence-awareness in Asterisk
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly simple application running on user desktop and
having just 3-4 buttons like
- online
- do not disturb
- forward to my mobile
and possibly also monitoring xscreensaver activity. This application
could then communicate with the * server (via AGI or SQL database or
something) and amend the dialplan accordingly.
Does anyone implemented it somewhere? How can I achieve this?
I am happy with just any hint pointing me to the right direction.
The implementation on the Asterisk side is quite easy.
Consider the case where you have
exten = 234,1,Dial(SIP/sip234)
Now you want to replace that with some kind of *-magic such that either
of the three options you mentioned can be selected.
exten = 234,1,GotoIf($[${DB(Status/${EXTEN})} = dnd]?10)
exten = 234,2,GotoIf($[${DB(Status/${EXTEN})} = away]?20)
exten = 234,3,Dial(SIP/sip234)
exten = 234,10,VoiceMail(b${EXTEN})
exten = 234,20,Dial( your mobile number)
(this is not beautiful, but you get the idea)
This way, any time someone calls the Asterisk database will be queried
for status information. You can put that information by hand from the
CLI ( database put Status 234 dnd ) or use some other means to set it. I
could imagine an Apache CGI script to do that, or you write a
proprietary (Windows,KDE,...) APP that runs in the user taskbar and is
able to somehow update the status in the Asterisk DB.
BTW you can set something in the asterisk DB from the shell with the
asterisk -rx database set. command.
HTH
Anselm
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Re: [asterisk-users] announcing inbound PSTN calls
Interesting!I think this can help for a start (but I don't know how to continue!!):[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav)
exten = _XXX,3,Dial(zap/1/${EXTEN})now, how to play the recorded message to the called party when he/she answers the phone?any help?On 11/10/06,
Jeronimo Romero [EMAIL PROTECTED] wrote:
I'm running asterisk 1.2.8. I would like PSTN inbound
calls to do the following:
1-once PSTN callers enter their desired extension; they have
to record their name
2-recording then announces that it is trying to locate the
user
3-asterisk calls local extension and announces callers
recorded name
4-local recipient user can choose to take the call, send it
to voicemail or transfer it to another extension
Is this possible in asterisk?? . If it is possible, what is
the name of this function? Is this documented anywhere?
What is the best approach to doing this?
Thanks in advance
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-- Gustavo BermanSysadminDepto. InformaticaUniversidad Nacional del ComahueCentro Regional Universitario Bariloche
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Re: [asterisk-users] announcing inbound PSTN calls
exten = _XXX,3,Dial(zap/1/${EXTEN},,A(somefile))
bails
Gustavo Berman wrote:
Interesting!
I think this can help for a start (but I don't know how to continue!!):
[incoming]
exten = s,1,Answer()
exten = s,2,Backgroud(enter-ext)
exten = _XXX,1,Playback(enter-name)
exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav)
exten = _XXX,3,Dial(zap/1/${EXTEN})
now, how to play the recorded message to the called party when he/she
answers the phone?
any help?
On 11/10/06, * Jeronimo Romero* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm running asterisk 1.2.8. I would like PSTN inbound calls to do
the following:
1-once PSTN callers enter their desired extension; they have to
record their name
2-recording then announces that it is trying to locate the user
3-asterisk calls local extension and announces callers recorded name
4-local recipient user can choose to take the call, send it to
voicemail or transfer it to another extension
Is this possible in asterisk?? . If it is possible, what is the name
of this function? Is this documented anywhere?
What is the best approach to doing this?
Thanks in advance
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--
Gustavo Berman
Sysadmin
Depto. Informatica
Universidad Nacional del Comahue
Centro Regional Universitario Bariloche
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Re: [asterisk-users] Presence-awareness in Asterisk
Hi Anselm,
Yes it looks promising.
somehow update the status in the Asterisk DB
and that's the problem - how can I access Asterisk DB remotely (in some
nice and elegant way)?
That's why I was more thinking about mysql - it is already running on my
* box and remote access is no problem.
Question is, if I could do the same trick you did with Asterisk DB with
Mysql.
Thanks!
Ondrej
P.S.
Apache cgi is a possibility, indeed.
Anselm Martin Hoffmeister wrote:
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly simple application running on user desktop and
having just 3-4 buttons like
- online
- do not disturb
- forward to my mobile
and possibly also monitoring xscreensaver activity. This application
could then communicate with the * server (via AGI or SQL database or
something) and amend the dialplan accordingly.
Does anyone implemented it somewhere? How can I achieve this?
I am happy with just any hint pointing me to the right direction.
The implementation on the Asterisk side is quite easy.
Consider the case where you have
exten = 234,1,Dial(SIP/sip234)
Now you want to replace that with some kind of *-magic such that either
of the three options you mentioned can be selected.
exten = 234,1,GotoIf($[${DB(Status/${EXTEN})} = dnd]?10)
exten = 234,2,GotoIf($[${DB(Status/${EXTEN})} = away]?20)
exten = 234,3,Dial(SIP/sip234)
exten = 234,10,VoiceMail(b${EXTEN})
exten = 234,20,Dial( your mobile number)
(this is not beautiful, but you get the idea)
This way, any time someone calls the Asterisk database will be queried
for status information. You can put that information by hand from the
CLI ( database put Status 234 dnd ) or use some other means to set it. I
could imagine an Apache CGI script to do that, or you write a
proprietary (Windows,KDE,...) APP that runs in the user taskbar and is
able to somehow update the status in the Asterisk DB.
BTW you can set something in the asterisk DB from the shell with the
asterisk -rx database set. command.
HTH
Anselm
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Re: [asterisk-users] announcing inbound PSTN calls
I would recommend you to record files with a uniqueid like var ${TIMESTAMP}[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Set(filename=${TIMESTAMP})
exten = _XXX,2,Record(/tmp/prompt${filename}:wav)exten = _XXX,3,Dial(zap/1/${EXTEN},A(${filename}.wav))On 11/10/06, bails
[EMAIL PROTECTED] wrote:exten = _XXX,3,Dial(zap/1/${EXTEN},,A(somefile))
bailsGustavo Berman wrote: Interesting! I think this can help for a start (but I don't know how to continue!!): [incoming] exten = s,1,Answer() exten = s,2,Backgroud(enter-ext)
exten = _XXX,1,Playback(enter-name) exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav) exten = _XXX,3,Dial(zap/1/${EXTEN}) now, how to play the recorded message to the called party when he/she
answers the phone? any help? On 11/10/06, * Jeronimo Romero* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to
record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to
voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere?
What is the best approach to doing this? Thanks in advance ___
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http://lists.digium.com/mailman/listinfo/asterisk-users -- Gustavo Berman Sysadmin Depto. Informatica Universidad Nacional del Comahue
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Re: [asterisk-users] announcing inbound PSTN calls
Am Freitag, den 10.11.2006, 00:07 -0500 schrieb Jeronimo Romero: I’m running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere? What is the best approach to doing this? In the voip-info.org wiki, there is an example for the usage of M (macro) for something similar to what you want to achieve http://www.voip-info.org/wiki/view/Asterisk+Cmd+Dial You might want to set a variable in the macro to some value if the option transfer or voicemail is set, so that in the main code you can have a GotoIf distinguishing those cases and doing the right thing(TM). HTH Anselm ___ --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] Presence-awareness in Asterisk
Am Freitag, den 10.11.2006, 16:33 +0100 schrieb Ondrej Valousek: Hi Anselm, Yes it looks promising. somehow update the status in the Asterisk DB and that's the problem - how can I access Asterisk DB remotely (in some nice and elegant way)? That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql. There I cannot help you. But - there is an Apache Manager API that can be used over the network: http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) It seems to have support for a DBPut command, which is what you need here. HTH Anselm ___ --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] Looking for IP phone / ATA that has builtin VPN support
I am looking for ip phone/ ATA that has built in VPN support. can any one suggest me any brand or customize firmware ? I think the Zultys 4x5's are supposed to have built in IPSec VPN support. Zultys was rumoured to be going out of business, though (not sure if that's really true). - Noah ___ --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] Question about Mitel phones
Hi all, Does anyone know if the Mitel phone features a webintreface for configuring the phone? Many thanks, Christian ___ --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] Outgoing problem on PRI
Mohamed A. Gombolaty wrote: Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=1-15,17-31 /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 Sounds like you need to fiddle around with your PRI Dialplan, test these out in zapata.conf. I am sure one of them will work for you. ;pridialplan=national ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;prilocaldialplan=national ; ; PRI callerid prefixes based on the given TON/NPI (dialplan) ; This is especially needed for euroisdn E1-PRIs ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 ;unknownprefix = ; ; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 ;unknownprefix = ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latest Debian and latest zaptel
On Fri, Nov 10, 2006 at 12:38:51AM +0100, Christian wrote: Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian I'm not sure what is the problem you have, but I'm waiting for a more detailed report. Could you replicate the problem? As I stated, the error looked strange to me, and I have asked you to provide me more details. Could you please do that? BTW: the harmless warnings you got are already fixed in the latest 1.4 SVN branch. Thanks russel and kpflemming. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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: Reg errors? Other anomalies? Check thosecapacitors!
On 9 Nov 2006, at 16:16, Ira wrote: At 05:00 AM 11/9/2006, you wrote: Several motherboard manufactures in the last 3-4 years have had capacitor problems, some reached the point of leaking others began to cause problems on the machine after they began to swell. Both Dell and IBM have replaced systems I know of and had the onsite techs check for swollen or leaking capacitors. I have an IBM where every single 470uf 25V cap on the board leaked at about 2.5 years. Replaced them all and it's still going strong. I think something went wrong in a capacitor plant somewhere a few years back and a whole bunch of bad ones got out in the wild. Yep, I had a VIA motherboard go bad last week, it started crashing every few days, then every few minutes. When I looked inside I could see that some of the caps had distinct 'domes' on the top where they should be flat. My supplier say it was out of warranty (3years old) but supplied me a new empty MB at a reducde cost, I swapped it in and was good to go. Lesson: Next time you open up a 3 year old server look at the electrolytic caps, if they look like alien is about to hatch from them, then start thinking about getting a replacement Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Modprobe Zaptel
On Thu, Nov 09, 2006 at 05:44:23PM -0800, Darryl Dunkin wrote: After running 'make install', do a 'depmod -a'. Then check /lib/modules for the file: find /lib/modules | grep zaptel Be sure the path /lib/modules/kernel/extra/zaptel.ko matches up with your currently running kernel (from uname-a) as that is where it will be checking. Also, as a test that is both harmless and can be run by non-root, use: /sbin/modinfo zaptel instead of 'modprobe zaptel' (explicit path is for the cse of non-root). If it shows you something, modprobe will know to locate module. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Modprobe Zaptel
On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote: Julian Varanini wrote: Hi, Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get module zaptel not found You need to edit /usr/src/linux/Makefile to remove make the EXTRAVERSION variable equal to -12 rather than the -12somethingsomethingsomething it is now. No need to recompile the kernel, just change the make file and recompile and reinstall zaptel. Is such horror normally needed with Mandrake? Doesn't Mandrake provide working kernel headers linked from /lib/modules/`uname -r`/build ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Voxee lag problems ?
I have noticed it too and do not use them anymore.. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ira Sent: Thursday, November 09, 2006 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voxee lag problems ? At 08:48 AM 11/9/2006, you wrote: Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem ? So it's not only me! Ira ___ --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 to get CDR to show answered calls only
Is there anyway to get CDR to show just the answered calls. Not by exporting to a spreadsheet and editing. We have ring groups and queues and CDR shows everything as calls received. Even if it's multiple extensions ringing it shows them as multiple calls received. This seems kind of goofy. ___ --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] Modprobe Zaptel
Tzafrir Cohen wrote: On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote: Is such horror normally needed with Mandrake? Doesn't Mandrake provide working kernel headers linked from /lib/modules/`uname -r`/build ? I have no issues with Mandriva 2006 or 2007 on compiling Zaptel, nor have I ever had to modify the Makefile. 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] Need to automatically park an incoming call and then connect to an extension.
Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro ___ --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] How to get CDR to show answered calls only
shadowym wrote: Is there anyway to get CDR to show just the answered calls. Not by exporting to a spreadsheet and editing. We have ring groups and queues and CDR shows everything as calls received. Even if it's multiple extensions ringing it shows them as multiple calls received. This seems kind of goofy. ___ --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 What I've come to realize is that CDR in asterisk is awful. We're looking at doing our own 'CDR' via the userfield ___ --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] Modprobe Zaptel
Doug Lytle wrote: Tzafrir Cohen wrote: On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote: Is such horror normally needed with Mandrake? Doesn't Mandrake provide working kernel headers linked from /lib/modules/`uname -r`/build ? I have no issues with Mandriva 2006 or 2007 on compiling Zaptel, nor have I ever had to modify the Makefile. 2006 and 2007 MAY have changed that. I needed it on 8.1 and 9.2 ___ --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] [resolved] asterisk 1,4 and google talk
Mani, I've gotten the same result both dialing from a gtalk client to SIP, as well as an SIP call to gtalk. You can run a jabber debug before the call is placed to see more debug info on what's causing the crash. With the module in Beta, I believe it's just a bug that needs to be worked out. Below you'll see the output of one of my calls. :M sysmast01*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/asterisk4273D1E7 type=set id=35 from=[EMAIL PROTECTED]/Talk.v1001EE54E14session type=initiate id=2077360010 initiator=[EMAIL PROTECTED]/Talk.v1001EE54E14 xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= sysmast01*CLI JABBER: gtalk_account INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq sysmast01*CLI *** glibc detected *** /usr/sbin/asterisk: munmap_chunk(): invalid pointer: 0xb7e47b73 *** === Backtrace: = /lib/libc.so.6(cfree+0x1bb)[0x9b667b] /usr/lib/asterisk/modules/chan_gtalk.so[0x82bde5] /usr/lib/asterisk/modules/chan_gtalk.so[0x82c436] /usr/lib/libiksemel.so.3(iks_filter_packet+0x129)[0x278789] /usr/lib/asterisk/modules/res_jabber.so[0x4000c7] /usr/lib/libiksemel.so.3[0x276b55] /usr/lib/libiksemel.so.3(iks_parse+0x5c1)[0x274ad1] /usr/lib/libiksemel.so.3(iks_recv+0x98)[0x276488] /usr/lib/asterisk/modules/res_jabber.so[0x3fbd70] /usr/sbin/asterisk[0x80eadfb] /lib/libpthread.so.0[0xac03db] /lib/libc.so.6(clone+0x5e)[0xa1a06e] Mani Sridhar wrote: hi, it turns out that the iksemel library (which i installed using an rpm) was returning 0 when the function iks_has_tls() was called. it should return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by running a test program i wrote, that calls iks_has_tls . it returned 0. i downloaded iksemel source, compiled it and now the test program returned 1. now, jabber show connected shows the google talk account as connected, but i don't see this buddy online on my other google talk buddy list. i added an extension in extensions.conf that calls Gtalk/buddy, and as soon as i call this extension, asterisk terminates due to a segmentation fault. it didn't seem like a core was dumped - i'm still looking for it. thanks sridhar _ Live the life in style with MSN Lifestyle. Check out! http://content.msn.co.in/Lifestyle/Default ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server
I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when asterisk starts saying the digits from the extension, the sound starts becoming very choppy. The voice after the digits is still choppy. Does anyone have a suggestion? The codec that asterisk is using with the softphone I am using is the GSM codec. Please advise, Mario ___ --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] Presence-awareness in Asterisk
Ciao Ondrej, That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql. Of course you can. In asterisk-addons there's the app MYSQL(), that does exactly what you want. See http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL for more details. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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 Max TNT SIP Authentication Issue, WORKING
Hi All, Thanks for your replies and help, I have this working now, TNT 11.0.6 and Asterisk 1.2.9.1, passing TNT SIP calls to Asterisk just fine. Working through the solution was extremely painful, took a week in the lab to figure out that I had my head shoved so far up my ass, I was eating lunch twice. Clarity of sight is infinitely more achievable with head dislodged from rectum. My lab setup simulated my production system cluster, with extension dialing through DUNDi look ups and multiple registration servers using Realtime Database for the User Agent authentication. It gets complicated. I setup the TNT between two registration servers, pri to one and sip to the other. Not like I would in production but hey, I was doing proof of concept testing. Going from PRI to TNT to SIP to Asterisk, the CID number was coming through to the the Asterisk server. The Asterisk server was translating the CID number into a user, then checking the Realtime Database for authentication info which it did find, but the call had none, so Asterisk dropped the call. As soon as I changed the CID number on the test phone to a 10 digit number, to simulate a call coming in from the PSTN, Asterisk did not find the number in the database and allowed the call to come in un-authenticated. The first usergroup reply from Barry asked about user=1239, this should have made me ask the question, why the asterisk server was seeing user 1239, but I was hung up on user=phone in the invite message and totally missed the correlation between the CID number translating to a user. The first lesson is to setup the lab to simulate real-world testing. I am curious why Asterisk inturprets the CID number as a user? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Stable clock with 2.6 and without Digium hardware.
On Fri, Nov 10, 2006 at 03:28:09PM +0200, Anton Tinchev wrote: Zoa wrote: Can you tell us how you do the testing ? 3-4 different ways. All gives same results, so test are pretty valid. 1. Interrupt counting inside the PC. 2. TDMoE packet counting on the switch. 3. External TDMoE equipment connected thru extreme network swich. The card of the PC and the device only connected to the switch. The switch filters all packets except TDMoE to de device. Calibrated oscilloscope conected to the interupt leg of the network chip. All coalescing/etc disabling. 4. Diagnostic results from firmware of the device. 5. ToDo test - oscilloscope directly conected to pads inside the PC, but needs mechanical work for each platform/type. 6. ToDo test - some driver relays the clock to simple hardware card in the PC and oscilloscope connected to it. How about meassuring it directly? For starters, take a look at zttest.c . (Though it could use some slightly better accuracy). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Need to automatically park an incoming call and then connect to an extension.
show applications in the CLI is your friend. Look for parkandannounce On 11/10/06, Mauro Zanin [EMAIL PROTECTED] wrote: Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro ___ --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] Realtime sippeers using NAT
I'm running sippeers and sipusers in my extconfig, and everything runs perfectly when a client is registered (ex. registers to port 1000), but when it re-registers the client is set to port 5060. This behavior does not take place if I use the static files. Both in my sip_buddies table for db, and sip.conf for static I have host=dynamic and nat=yes. :M ___ --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] Stable clock with 2.6 and without Digium hardware.
On Fri, Nov 10, 2006 at 03:04:46PM +0200, Anton Tinchev wrote: Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. Which card, BTW? 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable 1000 Hz for some time, but in moments stops/misses interrupts/goes away from 1000 Hz 2.6 with ztdummy USB_UHCI - don't works, needs some tweaking. Somebody knows good patch for it? 2.6 with ztdynamic as primary clock sources - some issues with 2.6 (ztdynamic not ported well to 2.6?) with the mainstream versions, somehow patches solves it. 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. Why does preemption conflict with HZ=1000 ? Totally disable preemption? On what system(s)? 2.4 with digium card - stable 1000 Hz 2.4 with ztdummy UHCI - stable 1000 Hz 2.4 with ztdynamic clock source - stable 1000 Hz/Depends on network conditions. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Returncode from command
Hi, How can I use a command return code in my dialplan? Example, I want to use the system command to run a perl script. This script exists with a code that I need to use in my dialplan. But I can figure out how to extract this value. Thanks for any pointers. Andre Courchesne ___ --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: IAX2 goes one way audio when lag gets bad
if anyone has one-way audio issues with iax over jittery connection, please look at bug report, what I created yesterday and report your experiences, I think this is one of the most serious bug, that must be identified and resolved before 1.4 will be released, thanks http://bugs.digium.com/view.php?id=8325 PJ Benjamin Jacob wrote: Martin Joseph wrote: On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said: Hi Matt - I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. On the customer's end I have the following config in iax.conf: trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and minexcesbuffer don't do anything. They are ignored by 1.2.x unless you specify that you want to use the old 1.0.x jitterbuffer. Instead you might try the parameters maxjitterbuffer, resyncthreshold, and maxjitterinterps. For more, you can check out the sample iax.conf. I believe, also, that you are correct in setting trunk=no. I know in the 1.0.x jitterbuffer, trunk was not fully supported. I think this is still the case with the 1.2.x jitterbuffer. If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... Marty seeing this thread a lil too late, i guess. So, am sorry if I am repeating things. When I was setting up my iax2 configs, I too had one way audio initialy. Tried the softphone on two machines(which incidentaly had asterisk running on them as well), to no avail. When I looked at the tcpdump on my asterisk server, I could see no rtp coming in from the two said machines. So, I shifted the softphone to another machine, this time on a windows machine, n voila! it worked like a charm. So, I hope you did have a look at the tcpdump to check on the rtp flow. cheerz - Ben. ___ --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] Harris picking up before extension
Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as soon as it reaches the Harris, eventhough the given extension can still be ringing. If I let the extension ring for a while and then pick up I only hear the prompt in the middle (or as far as it went till I picked up). Have tried kewlstart, loopstart, groundstart and even the answeronpolarityswitch configs in zapata.conf but can't find the solution. Any one having solved this problem?Alyed ___ --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] app_pppd - Could not read send data
Hi, did anyone managed to get chan-capi and app_pppd to work? Incoming call is accepted, pppd started, but no data transfered to pppd. I used app_pppd-060822.tgz, chan-capi 0.7.1, asterisk 1.2.13. Error messages: chan_capi.c:918 local_queue_frame: Could not write to pipe for ISDN1#0 DEBUG[3364] app_pppd.c: Could not read send data: Input/output error DEBUG[3364] app_pppd.c: Cancelling threads DEBUG[3364] app_pppd.c: pthread_join(info-thread_run, NULL) returned 0 -- Stefan Tichy [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] Harris picking up before extension
Alyed Tzompa wrote: Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as soon as it reaches the Harris, eventhough the given extension can still be ringing. If I let the extension ring for a while and then pick up I only hear the prompt in the middle (or as far as it went till I picked up). Have tried kewlstart, loopstart, groundstart and even the answeronpolarityswitch configs in zapata.conf but can't find the solution. Any one having solved this problem? Analog FXO ports are considered answered as soon as dialing is finished. You can switch to a non-analog port or loop your outgoing sound files. ___ --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] Queues and Timeouts.
Thomas Kenyon wrote: On a slightly different tone, has anyone written a queue viewer that runs as a daemon and serves the pages to the viewer rather than creates a manager login/logout event every few seconds? (If not I'll write one myself, but worth checking first). Just written one, so not important now. ___ --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] WIFI phones on asterisk
I have used WIP300, hitachi 5000 wireless phones on asterisk and have had good success. However, I am looking for a WIFI phone with integrated belt clip. Has anyone found any? I have tried after market clips and holders and those just don't work. THanks for sharing if someone has found something that works with asterisk. 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] WIFI phones on asterisk
On Fri, Nov 10, 2006 at 03:45:18PM -0500, Jerry Geis wrote: I have used WIP300, hitachi 5000 wireless phones on asterisk and have had good success. However, I am looking for a WIFI phone with integrated belt clip. Has anyone found any? I have tried after market clips and holders and those just don't work. You are very unlikely, IMO, *to* find anything: doing a *proper* belt clip, that can deal with the massive variations in belt sizes and thicknesses, and having a rugged enough primary housing to attach it to, are both major problems which militate against permanently mounted clips -- at least at this point in the lifecycle of Wiphones. And, to be clear: I haven't seen anything anywhere, no. 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
[asterisk-users] config template for Grandstreams
I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? I guess the procedure is to modify the template, then run the configuration tool on the template to generate the specific downloadable file..? Thanks Todd ___ --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] Stable clock with 2.6 and without Digium hardware.
snip Anton wrote: 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. How are you doing this? I saw a couple developers talk about rewriting ztdummy yo use the new hi-res kernel timers (kernel2.6.17), but did not notice any patches as a result. I've been pretty happy with the RTC-based ztdummy, except that the RTC can be a bit of a pig when run at 8192 (~8% cpu load). Hi-res timers firing every 1ms should produce a lighter load and be more accurate 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
Re[2]: [asterisk-users] Latest Debian and latest zaptel
me some tips on how to get it working and installed? Many thanks, Christian I'm not sure what is the problem you have, but I'm waiting for a more detailed report. Could you replicate the problem? As I stated, the error looked strange to me, and I have asked you to provide me more details. Could you please do that? BTW: the harmless warnings you got are already fixed in the latest 1.4 SVN branch. Thanks russel and kpflemming. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] Voxee lag problems ?
Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...On 11/10/06, Jonathan Borden [EMAIL PROTECTED] wrote:I have noticed it too and do not use them anymore.. Jon-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of IraSent: Thursday, November 09, 2006 11:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Voxee lag problems ?At 08:48 AM 11/9/2006, you wrote: Anyone having problems with voxee since last few days or is it justme ? In peek hours i get LAGGED when i do a iax2 show peers or even1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or evenLAGGED( also shows high in peak time even when no high latency ).No problems with any other provider . Anyone else having same problem ? So it's not only me!Ira___--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 ___ --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[2]: [asterisk-users] Latest Debian and latest zaptel
/libtonezone.so.1 /bin/ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -z -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then restorecon -v /usr/lib/libtonezone.so; fi /bin/sh: line 0: [: missing `]' /usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h /usr/bin/install: cannot create regular file `/usr/include/zaptel/tonezone.h': No such file or directory make: *** [install-include] Error 1 I really hope that you are able to tell me what the problem is why it doesn't want to install. Have been sitting with this for allmost three days now! All the best and many thanks, Christian On 2006-11-10 at 19:04 Tzafrir Cohen wrote: On Fri, Nov 10, 2006 at 12:38:51AM +0100, Christian wrote: Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian I'm not sure what is the problem you have, but I'm waiting for a more detailed report. Could you replicate the problem? As I stated, the error looked strange to me, and I have asked you to provide me more details. Could you please do that? BTW: the harmless warnings you got are already fixed in the latest 1.4 SVN branch. Thanks russel and kpflemming. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] monitor-join does not seem to work.
Despite of monitor-join being equal yes, I get individual -in and -out files for queue calls. My box runs Asterisk 1.2.10 and I've set up real-time queues. Does anybody have any idea of what is going on? Thanks in advance. Carlos. ___ --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[3]: [asterisk-users] Latest Debian and latest zaptel
) ; then restorecon -v /usr/lib/libtonezone.so; fi /bin/sh: line 0: [: missing `]' /usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h /usr/bin/install: cannot create regular file `/usr/include/zaptel/tonezone.h': No such file or directory make: *** [install-include] Error 1 I really hope that you are able to tell me what the problem is why it doesn't want to install. Have been sitting with this for allmost three days now! All the best and many thanks, Christian On 2006-11-10 at 19:04 Tzafrir Cohen wrote: On Fri, Nov 10, 2006 at 12:38:51AM +0100, Christian wrote: Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian I'm not sure what is the problem you have, but I'm waiting for a more detailed report. Could you replicate the problem? As I stated, the error looked strange to me, and I have asked you to provide me more details. Could you please do that? BTW: the harmless warnings you got are already fixed in the latest 1.4 SVN branch. Thanks russel and kpflemming. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] (no subject)
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036, 20001, 2727}
---
on the server side
same thing
plus
voip_users = ip from where i am connecting
--
can't seem to find anything else that should be opened on either side
to allow connection
--
i guess, help ?
--
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Re[3]: [asterisk-users] Latest Debian and latest zaptel
/xpd_fxs: 'dump_slic_cmd' exported twice. Previous export was in /root/zaptel-1.4.0-beta2/xpp/xpd_fxo.ko make[2]: Leaving directory `/usr/src/linux-headers-2.6.17-2-686' make[1]: Leaving directory `/root/zaptel-1.4.0-beta2' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ /usr/bin/install -c -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ /usr/bin/install -c -m 644 wct4xxp/*.ima /lib/firmware; \ fi Installed firmware /usr/bin/install -c -D -m 755 libtonezone.a /usr/lib/libtonezone.a /usr/bin/install -c -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 if [ -z -a `id -u` = 0 ]; then \ /sbin/ldconfig || : ;\ fi rm -f /usr/liblibtonezone.so /bin/ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 /bin/ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -z -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then restorecon -v /usr/lib/libtonezone.so; fi /bin/sh: line 0: [: missing `]' /usr/bin/install -c -D -m 644 zaptel.h /usr/include/zaptel/zaptel.h /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h /usr/bin/install: cannot create regular file `/usr/include/zaptel/tonezone.h': No such file or directory make: *** [install-include] Error 1 I really hope that you are able to tell me what the problem is why it doesn't want to install. Have been sitting with this for allmost three days now! All the best and many thanks, Christian On 2006-11-10 at 19:04 Tzafrir Cohen wrote: On Fri, Nov 10, 2006 at 12:38:51AM +0100, Christian wrote: Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian I'm not sure what is the problem you have, but I'm waiting for a more detailed report. Could you replicate the problem? As I stated, the error looked strange to me, and I have asked you to provide me more details. Could you please do that? BTW: the harmless warnings you got are already fixed in the latest 1.4 SVN branch. Thanks russel and kpflemming. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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: Re[3]: [asterisk-users] Latest Debian and latest zaptel
On Nov 10, 2006, at 11:12 PM, Christian wrote: Hi, But what is the problem, why doesnt it install? I am a little new to this so still learning. Many thanks, Christian Use the latest 1.4 svn version instead of beta2 That will probably fix your problem. I installed latest 1.4 svn checkout on debian etch 2 times today while playing with xen so I can confirm it's working great now. --- 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
Re: [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer
Everyone here is saying how it would be so great to have native desktop/outlook/exchange/etc support, but seriously, do you really think M$ is going to develop these products to use with the open source market? They're going to want to try monopolizing it and creating an environment where you need to use M$ VoIP products to take advantage of to try forcing users to buy their products like they do with everything else. On Tuesday 07 November 2006 05:28 pm, Dean Collins wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/as ia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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] Voxee lag problems ?
On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two... Now there is a definitive case of a 'lagged' communication channel! :-) Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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: Monitor, MixMonitor and volume levels
Steve Davies wrote: *bump* No suggestions at-all? Does anyone use this facility in a similar way and NOT have problems? Check the gain on your ISDN interface. The monitor command doesn't modify the volume by default. Have you tested calls via IAX to your cell? Leo ___ --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] Microsoft will enter VoIP market in earnest nextyear, says Ballmer
Nope I don't think for a moment they are going to encourage us to integrate by making it easy. This is why we need to develop more and more features (like the weather app - you know you can ftp to text to voice any file right?) The more features the more reason people will want to go with asterisk. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brodie Macleod Sent: Friday, 10 November 2006 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer Everyone here is saying how it would be so great to have native desktop/outlook/exchange/etc support, but seriously, do you really think M$ is going to develop these products to use with the open source market? They're going to want to try monopolizing it and creating an environment where you need to use M$ VoIP products to take advantage of to try forcing users to buy their products like they do with everything else. ___ --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] Microsoft will enter VoIP market in earnest next year, says Ballmer
a) Can we agree to stop using M$? Yeesh.b) The only trick to getting Asterisk to work is that Exchange 2007 is using SIP over TCP instead of SIP over UDP. Interestingly (and I just found this out myself) a product called Express SIP Router automagically translates UDP to/from TCP. See this article on the subject: http://www.windowsitpro.com/Windows/Article/ArticleID/71313/71313.htmlI am fantastically excited about seeing an Asterisk/Ex07 integration. On 11/10/06, Brodie Macleod [EMAIL PROTECTED] wrote: Everyone here is saying how it would be so great to have nativedesktop/outlook/exchange/etc support, but seriously, do you really think M$is going to develop these products to use with the open source market?They're going to want to try monopolizing it and creating an environment where you need to use M$ VoIP products to take advantage of to try forcingusers to buy their products like they do with everything else.On Tuesday 07 November 2006 05:28 pm, Dean Collins wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/as ia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial).___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -m+b ___ --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[5]: [asterisk-users] Latest Debian and latest zaptel
Hi, OK sounds great. How do I install that svn you are refering to? Sorry for all these newbie questions, but just want to get started and learn. Many thanks, christian On 2006-11-10 at 23:51 Michiel van Baak wrote: On Nov 10, 2006, at 11:12 PM, Christian wrote: Hi, But what is the problem, why doesnt it install? I am a little new to this so still learning. Many thanks, Christian Use the latest 1.4 svn version instead of beta2 That will probably fix your problem. I installed latest 1.4 svn checkout on debian etch 2 times today while playing with xen so I can confirm it's working great now. --- 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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] (no subject)
Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in pf.conf
# SIP (TCP)
voip_tcp = 5060
# SIP, IAX2, IAX, RTP, MGCP (UDP)
voip_udp = {5060, 4569, 5036, 20001, 2727}
---
on the server side
same thing
plus
voip_users = ip from where i am connecting
--
can't seem to find anything else that should be opened on either side
to allow connection
--
i guess, help ?
--
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [asterisk-users] WIFI phones on asterisk
I am surprised that you have had good success perhaps you haven't done proper testing?On 11/10/06, Jerry Geis [EMAIL PROTECTED] wrote:I have used WIP300, hitachi 5000 wireless phones on asterisk and have had good success.However, I am looking for a WIFI phone with integratedbelt clip. Has anyone found any?I have tried after market clips and holders and those justdon't work. THanks for sharing if someone has found somethingthat works with asterisk.Jerry___--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] SPA-941 (and others ) Transmit Sound Quality
Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [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] Dialing from Placed Calls on Polycom IP501 doesn't always work
Greetings, Has anyone noticed that attempting to place a call from the Placed Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? It is not related to the number being dialed, as we have observed two entries for the same number, one of which worked and the other didn't. We've experimented with calls that weren't answered at all, calls that were terminated by the caller and calls terminated by the recipient with no discernible pattern. Regards, CP ___ --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] Voxee lag problems ?
Add me to the list. Not only lagged, but also failures to register. AND, apparantly Paypal won't automatically authorize payments to them anymore. I'm not recharging my account anymore. On 11/10/06, Tim Panton [EMAIL PROTECTED] wrote: On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...Now there is a definitive case of a 'lagged' communication channel! :-)Tim Pantonwww.mexuar.netwww.westhawk.co.uk/___--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] Push to Talk settings.
Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you at cc because i know you find this interesting too and maybe meanwhile you already know more about this... Let's begin a thread. Thank you again. ___ --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[5]: [asterisk-users] Latest Debian and latest zaptel
Hi, Well, I managed to find out what svn was and I also downloaded the latest 1.4 version of zaptel and Asterisk. svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 But the problem is still there. Cannot install Zaptel. Asterisk and libpri installs just fine. Want to use ztdummy. Soon giving up on this one! Many thanks, Christian On 2006-11-10 at 23:51 Michiel van Baak wrote: On Nov 10, 2006, at 11:12 PM, Christian wrote: Hi, But what is the problem, why doesnt it install? I am a little new to this so still learning. Many thanks, Christian Use the latest 1.4 svn version instead of beta2 That will probably fix your problem. I installed latest 1.4 svn checkout on debian etch 2 times today while playing with xen so I can confirm it's working great now. --- 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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] app_swift: Failed to set voice
I'm trying to get app_swift (v0.9.1 from http://www.loopfree.net/app_swift/) working, but it's having issues (see below). I'm running 1.4.0beta3 on FC6. Any thoughts? *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/spa3k-fxs-08e884b0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Swift(SIP/spa3k-fxs-08e884b0, Diane^your text here!) in new stack [Nov 10 23:40:43] ERROR[21132]: app_swift.c:240 swift_exec: Failed to set voice. -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/spa3k-fxs-08e884b0, ) in new stack == Spawn extension (internal, 100, 3) exited non-zero on 'SIP/spa3k-fxs-08e884b0' Sent via the WebMail system at mail.valcom.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: Re[5]: [asterisk-users] Latest Debian and latest zaptel
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel cd zaptel make clean ; make distclean ; sh configure ; make ; make install modprobe ztdummy see if those give you any errors From: Christian [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re[5]: [asterisk-users] Latest Debian and latest zaptel Date: Sat, 11 Nov 2006 04:40:26 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f6.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 10 Nov 2006 19:42:14 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id E822A2FC431;Fri, 10 Nov 2006 20:40:10 -0700 (MST) Received: from psmtp.com (exprod8mx48.postini.com [64.18.3.148])by lists.digium.com (Postfix) with SMTP id 836352FC427for asterisk-users@lists.digium.com;Fri, 10 Nov 2006 20:40:06 -0700 (MST) Received: from source ([81.216.65.12]) by exprod8mx48.postini.com([64.18.7.10]) with SMTP; Fri, 10 Nov 2006 19:40:25 PST Received: from localhost (localhost [127.0.0.1])by iggypop2.siwnet.net (Postfix) with SMTP id EE8A9205886for asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) Received: from iggypop2.siwnet.net (localhost [127.0.0.1])by iggypop2.siwnet.net (Postfix) with ESMTP id 7D2DC205881for asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) Received: from u (231.142.216.81.static.tab.siw.siwnet.net [81.216.142.231])(Authenticated sender: [EMAIL PROTECTED])by iggypop2.siwnet.net (Postfix) with ESMTP id 5006F20584Dfor asterisk-users@lists.digium.com;Sat, 11 Nov 2006 04:40:22 +0100 (CET) X-Message-Info: LsUYwwHHNt3GES/zhXTpUM1zfXR0iU5Qwpl6IbFbp8M= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com References: [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED] X-Mailer: Courier 3.50.00.09.1098 (http://www.rosecitysoftware.com) (P) X-Virus-Scanned: ClamAV using ClamSMTP X-DSPAM-Result: Spam X-DSPAM-Processed: Sat Nov 11 04:40:22 2006 X-DSPAM-Confidence: 0.6060 X-DSPAM-Probability: 1. X-DSPAM-Signature: 45554626975671671310483 X-DSPAM-Factors: 15, Content-Transfer-Encoding*quoted+printable, 0.99000,Date*2006+04, 0.99000, Received*Nov+2006, 0.99000,Content-Transfer-Encoding*printable, 0.99000,Content-Transfer-Encoding*quoted, 0.99000,From*Christian [EMAIL PROTECTED], 0.99000,Date*Nov+2006, 0.99000, Hi, 0.99000,Received*2006+04, 0.99000, both, 0.12167, what, 0.16866,com, 0.17363, up, 0.18386, Received*11, 0.19184, so, 0.19824 X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 11 Nov 2006 03:42:15.0633 (UTC) FILETIME=[6146DC10:01C70543] Hi, Well, I managed to find out what svn was and I also downloaded the latest 1.4 version of zaptel and Asterisk. svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 But the problem is still there. Cannot install Zaptel. Asterisk and libpri installs just fine. Want to use ztdummy. Soon giving up on this one! Many thanks, Christian On 2006-11-10 at 23:51 Michiel van Baak wrote: On Nov 10, 2006, at 11:12 PM, Christian wrote: Hi, But what is the problem, why doesnt it install? I am a little new to this so still learning. Many thanks, Christian Use the latest 1.4 svn version instead of beta2 That will probably fix your problem. I installed latest 1.4 svn checkout on debian etch 2 times today while playing with xen so I can confirm it's working great now. --- 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 __ NOD32 1861 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com
[asterisk-users] Soundfiles adding during phone calls
I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --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
