Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID
2009/7/7 Doug Lytle supp...@drdos.info Olivier wrote: Please, allow me to ask what is this Transmitting Station ID ? Google is you friend: http://encyclopedia.thefreedictionary.com/Transmitting+Subscriber+Identification Thanks ! I still have to improve my googling ! 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Gain Control
2009/7/7 Brent Davidson br...@texascountrytitle.com Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8 solves the problem with long-distance lines, but then local calls say the person in my office is too loud. I understand that it is going to be difficult to reliably detect a major drop in the volume at the far end of the call, but I'm just wondering if there is a good solution for this. We're using Rhino WC4-FXO-ec cards and the OSlec echo canceler (since the on-board echo canceler didn't seem to help our echo issues) hello, I'm afraid I can't be of any help but as a first step, I'm wondering if there's a way to translate user experience such as too loud into figures such volume is 8 ? Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] documentation of DAHDI dial options
2009/7/7 Jared Smith jsm...@digium.com On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote: I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like core show application Dial Does such thing exists? I don't think that such a thing exists. The only ones I'm aware of are: 1) Channel Groups. DAHDI/g1/5551212 dials 5551212 on the first available channel in group one, searching from lowest to highest DAHDI/G1/5551212 dials 5551212 on the first available channel in group one, searching from highest to lowest DAHDI/r1/5551212 dials 5551212 on the first available channel in group one, going in round-robin fashion (and remembering where it last left off), searching from lowest to highest DAHDI/R1/5551212 dials 5551212 on the first available channel in group one, searching in round-robin fashion from highest to lowest. 2) Distinctive ring DAHDI/4r1 dials channel 4 (presumably an FXS channel), and uses distinctive ring style one. If I recall, there are four different distinctive ring styles... so you could replace r1 with r2, r3, or r4. 3) Answer confirmation DAHDI/1c/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and not consider the call answered until the called party presses #. This is useful because of the way analog signaling works. Without this setting, Asterisk considers any outbound analog call on an FXO port answered just as soon as it has been dialed. 4) Digital calls DAHDI/1d/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and that it's a digital call. If I remember correctly, this is used for ISDN calls to set the bearer capability. I've taken a quick look in channels/chan_dahdi.c in TRUNK, and it seems to match up with my understanding, as I didn't see any other options stand out. While poking around in there, I found the following comment: /* * data is ---v * Dial(DAHDI/pseudo[/extension]) * Dial(DAHDI/channel#[c|rcadance#|d][/extension]) * Dial(DAHDI/(g|G|r|R)group#(0-63)[c|rcadance#|d][/extension]) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * rcadance# - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ That's probably as definitive an answer as you're going to get. What is this was commented such as it could be added to a core show application Dial ? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small site survivability
2009/7/6 Jonathan Thurman jthurma...@gmail.com We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). What happens for IT when WAN fails ? Are people still able to work or not ? If they are, then it should be possible to use current routers (if they have such POTS interfaces) as Media gateways and have a local resource to act as a backup Asterisk server. If they are not, having IT and Telephony to share the same backup WAN is advisable. Thanks for any suggestions! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Skype
This is not currently possible. Work in progress. -- Sent from mobile device On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] documentation of DAHDI dial options
Jared Smith schrieb: On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote: I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like core show application Dial Does such thing exists? I don't think that such a thing exists. The only ones I'm aware of are: 1) Channel Groups. DAHDI/g1/5551212 dials 5551212 on the first available channel in group one, searching from lowest to highest DAHDI/G1/5551212 dials 5551212 on the first available channel in group one, searching from highest to lowest DAHDI/r1/5551212 dials 5551212 on the first available channel in group one, going in round-robin fashion (and remembering where it last left off), searching from lowest to highest DAHDI/R1/5551212 dials 5551212 on the first available channel in group one, searching in round-robin fashion from highest to lowest. 2) Distinctive ring DAHDI/4r1 dials channel 4 (presumably an FXS channel), and uses distinctive ring style one. If I recall, there are four different distinctive ring styles... so you could replace r1 with r2, r3, or r4. 3) Answer confirmation DAHDI/1c/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and not consider the call answered until the called party presses #. This is useful because of the way analog signaling works. Without this setting, Asterisk considers any outbound analog call on an FXO port answered just as soon as it has been dialed. 4) Digital calls DAHDI/1d/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and that it's a digital call. If I remember correctly, this is used for ISDN calls to set the bearer capability. I've taken a quick look in channels/chan_dahdi.c in TRUNK, and it seems to match up with my understanding, as I didn't see any other options stand out. While poking around in there, I found the following comment: /* * data is ---v * Dial(DAHDI/pseudo[/extension]) * Dial(DAHDI/channel#[c|rcadance#|d][/extension]) * Dial(DAHDI/(g|G|r|R)group#(0-63)[c|rcadance#|d][/extension]) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * rcadance# - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ That's probably as definitive an answer as you're going to get. Thanks, great. So now we have what I was looking for. Now we need a place to make this documentation public. I wonder what could be a place for that? IMO it would be great if the documentation would be inside Asterisk. Maybe it could be added to core show channeltype dahdi. What do you think? Otherwise this information is again lost, and voip-info pages are always outdated. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Gain Control
17. Automatic Gain Control (Brent Davidson) Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? Have a look at asterisk-1.6.1 and module func_speex.so, which provides AGC function. This function can be applied to any channel. Documentation: http://www.voip-info.org/wiki/view/Asterisk+func+speex and *CLI core show function AGC Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Skype
DHAVAL INDRODIYA wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval Chan_celiax can apparently interface with a copy of the skype client running on the same machine, (I've not tried it so don't know how well it works). Other than that there is I gether an online SIP to Skype service (that someone will probably mention in a moment). As Alex suggests, digium are working on their own channel driver. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream 2010 and blinky lights
That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. 1) When the agent logs in / logs out, we rewrite the part of the dialplan for the hints and reload the dialplan 10 seconds after the *last* login / logout 2) For the MWI, we give each phone a fake voicemail (let's say _0001_). When an agent logs in, we link /var/spool/asterisk/voicemail/_0001_ to /var/spool/asterisk/voicemail/[mailbox] (where [mailbox] is the mailbox of the agent) and when they log out, we remove /var/spool/asterisk/voicemail/_0001_ This seems to work - the MWI lights up / off depending on the new vm within a couple of seconds 3) When checking for voicemail, each phone is configured to dial - the dialplan then checks the callerid (set by #1) and gets the mailbox for the agent. As I said, a bit of a hack, but it works for me ;) I know that this won't work for 1.6, but we are coming up with an alternative plan using Minivm Julian Andrew Thomas wrote: The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream 2010 and blinky lights
On 8/7/09 8:52 PM, Andrew Thomas wrote: That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. Why don't you just use func_devstate which was backported to 1.4? That way you can just set a DB variable on login/logout. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c [RESOLVED]
This is my jabber.conf : [general] debug=yes ;;Turn on debugging by default. ;autoprune=no ;;Auto remove users from buddy list. ;autoregister=yes ;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection serverhost=openfire.jocan.local ;;Route to server for example talk.google.com username=aster...@openfire.jocan.local ;;Username with optional roster. secret=password ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. timeout=100 ;;Timeout on the message stack. What did I change ? - I use the FQDN for the arguments 'serverhost' and 'username' in stead of the IP-address of the OpenFire-server. Now all works well : jabber show connected Jabber Users and their status: User: aster...@openfire.jocan.local - Connected Number of users: 1 Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Skype
when using sisky you could integrate an ivr menu Alex Balashov schreef: This is not currently possible. Work in progress. -- Sent from mobile device On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groet Kind Regards, Mit den besten Grüßen, Fons van der Beek, 84-IT BV http://www.84-it.com/index.php?option=com_contentview=articleid=2Itemid=2 T +31 475 769002 M +31 6 29296243 E fons.vanderb...@84-it.com mailto:fons.vanderb...@84-it.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk addon mysql - is mysql connection persistent
Thanks Miguel Molina :) I was bit curious about that as I am using few asterisk boxes connected to a mysql server. And that mysql server sometimes gets lots of connections from other sides ( other than asterisk boxes) . So if asterisk-mysql holds dedicated persistant connection , it means cdr are being pushed to database as in normal way. On Wed, Jul 8, 2009 at 5:27 AM, Miguel Molina mmol...@millenium.com.cowrote: Shahid Tel escribió: Hi Guys, As it looks like from CLI command show cdr mysql , can somebody confirms that cdr-mysql creates persistent connection with in asterisk? show cdr mysql shows connected to u...@dbhost from 18 hours ... Yes, the MySQL CDR addon creates a persistent connection to the database. If the database server goes down, the addon tries to reconnect so if it succeeds no records are lost or only a few, I'm not sure. The addon won't die, neither asterisk. For example: cdr mysql status CLI command shows me this: Connected to user@IP, port 3306 using table table for 19 days, 19 hours, 32 minutes, 2 seconds. Wrote 4256045 records since last restart and 294847 records since last reconnect. Restart is the last asterisk restart. Last reconnect, is the last time the connection went down and reconnected because the server went down or you killed the connection from the MySQL monitor. So if for any reason you need to do a quick restart of MySQL, you won't lose CDR records if no calls are hungup during the MySQL restart cycle. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote: Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards Whether true or not, I was told that nearly 80% of people in the US have caller ID. I would say that number is much higher for business, especially on PRI circuits. I think the two big motivators there were packaging of services, for X amount extra, you get caller ID, call waiting, voicemail on at the telco, etc The other factor was the proliferation of telemarketing. Before the DNC, a white pages listed home phone could ring a dozen times a day by people selling stuff. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk + cisco as5400 t.38 fax sending.
Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38 through asterisk to a PST gateway that supports t.38 too. Is that true ? If so, what elements you need to make it work beside asterisk and the PSTN trunk ? Thanks all.- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
On 7/8/09, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote: Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards Whether true or not, I was told that nearly 80% of people in the US have caller ID. I would say that number is much higher for business, especially on PRI circuits. I think the two big motivators there were packaging of services, for X amount extra, you get caller ID, call waiting, voicemail on at the telco, etc The other factor was the proliferation of telemarketing. Before the DNC, a white pages listed home phone could ring a dozen times a day by people selling stuff. -- Thanks, Steve Totaro In Canada, their telephone network is set up to allow for dynamic CallerIDname on PRIs just like how CallerIDnumber works here in the USA. We didn't believe it at first until we tried it, but they seem to be the only country we've worked in, out of a few dozen countries, that allows dynamic CallerIDname defined on a per-call basis. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calculate data traffic
To calculate the monthly data traffic that is generated by VoIP-calls, is it as simpel as 80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month = 585.9375 MB traffic / month ??? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SALE 70% OFF on Pfizer
Title: asterisk-users@lists.digium.com Wed, 8 Jul 2009 03:34:15 +0100 New from WebMD: Dear asterisk-us...@lists.digium.com!receipt confirmation Sign-up today! You are subscribed as asterisk-us...@lists.digium.com. View and manage your WebMD newsletter preferences. Subscribe to more newsletters. Change/update your email address. WebMD Privacy PolicyWebMD Office of Privacy1175 Peachtree Street, Suite 2400, Atlanta, GA 30361© 2009 WebMD, LLC. All rights reserved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor/Queue and Tranfers
Thanks for the reply.
1. The extensions in the Queues are setup as Agent members, defined in
Agents.conf, then within the definition of the queue in queues.conf
they are made members of the queue.
2. As for the recording my diaplan is as follows:
[main-line]
exten = s,1,NoOp()
exten = s,n,NoOp(CallerID-dnid ${CALLERID(dnid)}))
exten = s,n,NoOp(CallerID-number ${CALLERID(number)}))
exten = s,n,NoOp(CallerID-name ${CALLERID(name)}))
exten = s,n,Wait(2)
exten = s,n,Answer
exten = s,n,Playback(/var/lib/asterisk/sounds/custom/queue_greeting)
exten =
s,n,MixMonitor(/var/www/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(number)}_${UNIQUEID}.WAV||)
exten = s,n,GotoIfTime(20:00-7:45|mon-fri|*|*?Afterhours)
exten = s,n,GotoIfTime(*|sat-sun|*|*?Afterhours)
exten = s,n,GotoIfTime(*|*|25-26|dec?Afterhours)
exten = s,n,GotoIfTime(*|*|1|jan?Afterhours)
exten = s,n,GotoIfTime(*|*|1|sep?Afterhours)
exten = s,n,GotoIfTime(*|*|21|mar?Afterhours)
exten = s,n,GotoIfTime(17:30-20:00|*|10|apr?Afterhours)
exten = s,n,GotoIfTime(*|*|11|nov?Afterhours)
exten = s,n(Businesshours),Queue(MainQueue|t|||3600)
exten = s,n,Hangup
exten = s,n(Afterhours),Queue(AFTERHOURS|t|||3600)
exten = s,n,Hangup
I am under the impression that MixMonitor records both streams and
mixes them at the same time, meaning I'm not recording on the caller
or callee but both. However, I could be mistaken. Thanks.
On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co wrote:
Darrin Henshaw escribió:
2. The issue does seem to be limited to MixMonitor and the Queue
application, as in testing I setup mixmonitor on my extension dialed it from
outside the company(my cell phone) and transferred the call without stopping
the recording.
I have a couple of questions on this:
1. Are you using SIP/IAX2/whatever extensions as queue members or Agent type
members?
2. If you are using Agent members, on the queued calls (though is the same
call) are you recording from the Agent channel (callee) or from the client
channel (caller)? That would make a difference in case of a transfer, because
the callee leg changes but the caller leg is the same.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Automatic Gain Control
If you are using a large number of DAHDI channels, you could designate a chunk of them as non-local since you can control RXGAIN on each channel. You would have to work out something with your TELCO since your'e a dead duck control-wise once you answer the call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Tuesday, July 07, 2009 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Automatic Gain Control Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8 solves the problem with long-distance lines, but then local calls say the person in my office is too loud. I understand that it is going to be difficult to reliably detect a major drop in the volume at the far end of the call, but I'm just wondering if there is a good solution for this. We're using Rhino WC4-FXO-ec cards and the OSlec echo canceler (since the on-board echo canceler didn't seem to help our echo issues) Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking with ISDN
The sort of trunk does matter; I don't know about ISDN, but I get different behavior on DAHDI vs SIP, so that's one verification that you are dealing with a necessarily fixed set of values. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm Sent: Tuesday, July 07, 2009 2:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call parking with ISDN Since no one has responded to this, I am wondering if there are two kinds of call park. I haven't worked with European ISDN, but if it has a call park feature, that would be distinctly different from the Asterisk PABX call park feature. The Asterisk feature should not matter what sort of trunk was involved, which is why I am wondering. On the other hand, if there is an ISDN park, I'm not sure Asterisk would support it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play a recorded message when a fax is detected ?
You should initiate a second call or send a voicemail. You don't want to
mess too much with what is working.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, July 07, 2009 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Play a recorded message when a fax is detected ?
Hi,
I'm configuring a system so that end user can receive phone and calls using
the same extension and DID.
At the moment, fax are correctly detected but I'm trying to improve end user
experience.
Relevant dialplan (from extensions.ael) is :
fax = {
Verbose(0,Incoming fax from ${CALLERID(num)});
FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif;
ReceiveFAX(${FAXFILE});
HangUp();
};
What I would to improve is when a fax is detected, instead of hanging up the
receiving extension, play a recorded message like you're receiving a fax
(if receiving end is human, or nothing at all if it's a voicemail).
What would you advise me to try ?
Regards
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Re: [asterisk-users] Caller ID (name) - where does it come from?
CALLERID(name) is a TELCO specific field. In the long run, you will be best served using your own lookup of a database using CALLERID(num), since CID(name) is unreliable and in some cases costly. IMO, you would be well served with an app (AGI?) that recorded valid names into the database and let you insert the names where they aren't. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D. Hassler Sent: Tuesday, July 07, 2009 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID (name) - where does it come from? Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g.722 + loudness
Hose wrote: Hi, We've been running g.722 in asterisk 1.6.09 for awhile now, with a PRI and numerous sip clients. Internal sip to sip and sip to pri (and vice versa) work fine between g.722 and ulaw - the transcoding is acceptable. The only time it fails is when we utilize a meetme conference bridge. With a Polycom IP 6000 + a call over the PRI, the person calling in over the PRI sounds distorted when they're barely talking at a normal volume. Anything over a normal volume results in terrible clipping. Bringing the volume down on the Polycom either via software settings or the actual volume keys doesn't stop the distortion, so that points to a problem with asterisk (the volume can be very loud, barely audible, but you can still hear the clipping occuring). By clipping, I mean the static that happens when you have a signal that's too loud. The thing is, when you call directly into the Polycom over the PRI, it's fine. This ONLY happens during a conference call with g.722, though this might be because asterisk is negotiating a ulaw connection when called direct from the PRI - is there a way to check what codec it's negotiated during the call? I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? I'm not sure in which version of Asterisk it was fixed, but there was a 6dB gain error in the G.722 codec until fairly recently. You are probably hitting that problem. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Restarting of B-channel on span 1
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI Channels
are restating after every interval of one hour, and i have search and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i am
not able to find it is listed some where} why this is nesessary? and if this is
not nessary how to stop it...
I think we all already know the message , but posting it for future reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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Re: [asterisk-users] g.722 + loudness
Hose wrote: I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restarting of B-channel on span 1
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i am
not able to find it is listed some where} why this is nesessary? and if this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
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Re: [asterisk-users] false answer on zaptel
On Mon, 06 Jul 2009 10:31:18 -0500 Brent Davidson br...@texascountrytitle.com wrote: Botond Botyanszki wrote: Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. What Telco are you using? Do you have callprogress=yes or hanguponpolarityswitch=yes in your zapata/dahdi .conf? No I didn't have them. These seemed to solve it, thanks a lot Brent! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP-1200 G.722?
Can anyone here have experience using G.722 on the Grandstream GXP-1200? It's supposed to support the codec, but I wonder if the handset does it justice? The older BT-200 also supported the codec, but the handset was not good enough. You could only hear the improved call quality using a headset. Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves FWD 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small site survivability
snip Audiocodes supports SRST on their mediapack analog gateways. This might be a viable option. I haven't used any Audiocodes devices before. Are people pleased with them? snip Deploy a lot of small asterisk based appliances... This way you can completely decentralise your setup and give each office it's own autonomous system, only needing the WAN links for inter-site calls (and maybe your backhaul to the PSTN) We do not want to decentralise our configuration. The whole point of pulling all of these sites together was to centralise management. We also have a lot of users that move to a different site every year and keep their DID as long as they are within the same county. We simply need some way to provide basic call management for local 911 access in the case of WAN failure. Our Cisco devices do this for any phone using SCCP. If you want to buy an additional license you can have SIP too... snip What happens for IT when WAN fails ? Are people still able to work or not ? I work in K-12 education, so while our users will complain that they don't have internet/email/etc, they continue to work with or without the WAN connection. Even if normal phone service is not available, we HAVE to provide 911 access. If they are, then it should be possible to use current routers (if they have such POTS interfaces) as Media gateways and have a local resource to act as a backup Asterisk server. I am trying to avoid adding additional servers at this small sites. Some sites are nothing more than a portable with Metro Ethernet connection and a fan-less router and switch. If they are not, having IT and Telephony to share the same backup WAN is advisable. Backup WAN links... I wish! Thanks for the input. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g.722 + loudness
Kevin P. Fleming wrote: Hose wrote: I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). It is not in the 1.6.0.10 Changelog nor the 1.6.1.1 Changelog. It is, however, in the 1.6.0.11-rc1 Changelog and the 1.6.2.0-beta3 Changelog. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling non-extension numbers issue
The two logs that I have been able to find are messages on the asterisk server in debug. Unfortunately, Nokia does not have any kind of logging (sucks). What I can see is that it is definitely a phone issue, just stuck on where to go from here. First, this if from asterisk in debug 1. -- Registered SIP '104' at 192.168.111.182 port 5060 expires 3600 -- Saved useragent E71-2 RM-346 200.21.118 for peer 104 -- Got SIP response 400 Bad Request back from 192.168.111.182 Second, these are two connection attempts to the same asterisk server, one successful, one not: 1. Successful: Found peer '103' Found RTP audio format 96 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 13 Peer audio RTP is at port 192.168.111.183:49152 Found unknown media description format AMR for ID 96 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 98 Found audio description format CN for ID 13 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.111.183:49152 Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: sip:1...@192.168.111.183 2. Failure: Found peer '104' - message ends at this point Third, this is from the IIS logs where the same devices connect FROM, one successful, one not: 1. Successful: 2009-07-01 15:41:10Local4.Info192.168.111.1 %PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35/5060 to inside:192.168.111.182 from INVITE message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from Response 4xx message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Via UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from ACK message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Via UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from INVITE message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35/5060 to inside:192.168.111.182 from INVITE message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from Response 100 message 2009-07-01 15:41:10Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from Response 180 message 2. Failure: 2009-07-01 15:44:53Local4.Info192.168.111.1 %PIX-6-607001: Pre-allocate SIP Via UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from INVITE message 2009-07-01 15:44:53Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35/5060 to inside:192.168.111.182 from INVITE message 2009-07-01 15:44:53Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Signalling UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from Response 4xx message 2009-07-01 15:44:53Local4.Info192.168.111.1%PIX-6-607001: Pre-allocate SIP Via UDP secondary channel for outside:67.220.106.35 to inside:192.168.111.182/5060 from ACK message It appears that the failure attempt does not contain the same Response 100/180 commands as the successful one. Anyway, if someone sees something obvious, please let me know, it would be greatly appreciated. We are stumped on this end and just really not sure how to proceed. -Kayton Hi, If You don't see anything on the command line of *, there might be an issue with Your phone settings. I don't know anything about the nokias, but I *think* it might be possible, that the phone connects to anything other than Your * box in case of the outbond number. AFAIK the * sends a 404-Error back on an non existing extension. In this case the phone would not show up a connection time-out. So I would check the settings on the phone. Or maybe You could do a network trace with tcpdump or ngrep to double check, that the phone really tries to connect to *. HTH, Karsten Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton
Re: [asterisk-users] Caller ID (name) - where does it come from?
Well, Teliax says they have no access to the PSTN's database, but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). On Wed, Jul 8, 2009 at 8:51 AM, Danny Nicholas da...@debsinc.com wrote: CALLERID(name) is a TELCO specific field. In the long run, you will be best served using your own lookup of a database using CALLERID(num), since CID(name) is unreliable and in some cases costly. IMO, you would be well served with an app (AGI?) that recorded valid names into the database and let you insert the names where they aren’t. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Barry D. Hassler *Sent:* Tuesday, July 07, 2009 12:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Caller ID (name) - where does it come from? Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor/Queue and Tranfers
Un-topposting...
On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co
wrote:
Darrin Henshaw escribió:
2. The issue does seem to be limited to MixMonitor and the Queue
application, as in testing I setup mixmonitor on my extension dialed it from
outside the company(my cell phone) and transferred the call without stopping
the recording.
I have a couple of questions on this:
1. Are you using SIP/IAX2/whatever extensions as queue members or Agent type
members?
2. If you are using Agent members, on the queued calls (though is the same
call) are you recording from the Agent channel (callee) or from the client
channel (caller)? That would make a difference in case of a transfer,
because the callee leg changes but the caller leg is the same.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
Darrin Henshaw escribió:
Thanks for the reply.
1. The extensions in the Queues are setup as Agent members, defined in
Agents.conf, then within the definition of the queue in queues.conf
they are made members of the queue.
2. As for the recording my diaplan is as follows:
[main-line]
exten = s,1,NoOp()
exten = s,n,NoOp(CallerID-dnid ${CALLERID(dnid)}))
exten = s,n,NoOp(CallerID-number ${CALLERID(number)}))
exten = s,n,NoOp(CallerID-name ${CALLERID(name)}))
exten = s,n,Wait(2)
exten = s,n,Answer
exten = s,n,Playback(/var/lib/asterisk/sounds/custom/queue_greeting)
exten =
s,n,MixMonitor(/var/www/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(number)}_${UNIQUEID}.WAV||)
exten = s,n,GotoIfTime(20:00-7:45|mon-fri|*|*?Afterhours)
exten = s,n,GotoIfTime(*|sat-sun|*|*?Afterhours)
exten = s,n,GotoIfTime(*|*|25-26|dec?Afterhours)
exten = s,n,GotoIfTime(*|*|1|jan?Afterhours)
exten = s,n,GotoIfTime(*|*|1|sep?Afterhours)
exten = s,n,GotoIfTime(*|*|21|mar?Afterhours)
exten = s,n,GotoIfTime(17:30-20:00|*|10|apr?Afterhours)
exten = s,n,GotoIfTime(*|*|11|nov?Afterhours)
exten = s,n(Businesshours),Queue(MainQueue|t|||3600)
exten = s,n,Hangup
exten = s,n(Afterhours),Queue(AFTERHOURS|t|||3600)
exten = s,n,Hangup
I am under the impression that MixMonitor records both streams and
mixes them at the same time, meaning I'm not recording on the caller
or callee but both. However, I could be mistaken. Thanks.
Well, you are recording (monitoring) from the caller channel, not the
Agent channel, so you should get the complete recording, even if it's
transferred (someone please correct me if I'm wrong). AFAIK, when we do
a attended transfer and the original Queue call is being recorded we end
up with two types of recordings, because I have MixMonitor into the
initial Queue and the transfer Queue too:
1. Initial call connected to agent - Conversation with second agent
(the time where the two agents talk to each other is not recorded
because the caller channel is put into MoH, and is not bridged at that
time).
2. Transferrer call connected to transfer queue and to second agent (the
caller is now the transferrer in this case). The conversation between
the agents is recorded with the resulting transferred call, where the
initial caller becomes the new caller of the transferred call when the
transferrer hangs up.
So for me there's no way to have the whole attended transfer process on
a same recording file (initial call, the conversation between the two
agents, and the resulting transferred call).
On the other hand, I see that you don't use the 'b' option in
MixMonitor, do you record the entire call including the MoH that the
caller hears before is connected to someone? Try the 'b' option, it only
records the call while it's bridged to another channel (i.e. talking to
another one), that can save you lots of disk space while you record the
relevant part of the call. You can see on the CLI the events where
MixMonitor starts/stops the recording, that will help you troubleshoot
your issue:
== Begin MixMonitor Recording SIP/TRUNK_SWITCH4PRI-b281a910
...
== End MixMonitor Recording SIP/TRUNK_SWITCH4PRI-b281a910
Hope it helps.
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Automatic Gain Control
Danny Nicholas wrote: If you are using a large number of DAHDI channels, you could designate a chunk of them as non-local since you can control RXGAIN on each channel. You would have to work out something with your TELCO since your'e a dead duck control-wise once you answer the call. Yuck. I could see that being a temporary workaround, but it is not a good permanent solution. And even as a workaround it wouldn't work for my application. Each of our remote offices normally only has 1 employee (2 at most) and 2 incoming lines in a rollover setup. I know I've probably asked this before but which parameters do txgain and rx gain control? I've heard conflicting explanations. Looking at it from a telco equipment standpoint I would say rxgain should be the gain on the sound received from the far end of the PSTN and txgain is the sound leaving the TDM card over the PSTN. But I've seen a couple of explanations say that rxgain sets the volume of sound flowing into the zap/dahdi module from other channels and that txgain sets the volume flowing out of the zap module to other modules. That would have the effect of reversing what seems like logical functions and make rxgain actually control the volume being sent out to the PSTN and txgain set the volume coming in from the PSTN. I have not had opportunity to run any tests to verify for myself which explanation is correct. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] q: install asterisk + asteris-gui
hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 /var/lib/asterisk/static_html now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is empty... what am i supposed to do now? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] q: which Browser-GUI do u guys use?
___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
- tom tomabr...@gmail.com wrote: hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 /var/lib/asterisk/static_html now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is empty... what am i supposed to do now? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 'make samples' from your Asterisk source dir --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: which Browser-GUI do u guys use?
- tom tomabr...@gmail.com wrote: *MY* browser must be experiencing problems. I thought you posted a message but it appears blank. /sarcasm I'm a huge fan of elinks. It's cross platform and works great. --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
shame on me...yes i had several different installations of asterisk, just to try it out. but i deleted everything before i went on installing a different version or vendor. so, make samples did the trick! i now have the missing files. thx (i didnt do it before coz somehow samples + freepbx) screwd my setup and nothing was working at the end)(thats as well the reason why i asked what u guys use as a gui) thx for everyone On Wed, Jul 8, 2009 at 3:12 PM, Danny Nicholas da...@debsinc.com wrote: Since /etc/asterisk is empty, you have either relocated your conf files or put them in a database. Assuming neither, just create manager.conf in /etc/asterisk with this setup [general] Enabled = yes Port = 5038 Webenabled=yes Bindaddr = 1.2.3.4 [loginname] Secret=secret And restart asterisk -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *tom *Sent:* Wednesday, July 08, 2009 1:50 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] q: install asterisk + asteris-gui hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 /var/lib/asterisk/static_html now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is empty... what am i supposed to do now? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
On Wed, 2009-07-08 at 14:49 -0400, tom wrote: - repointes apache /var/www/1234 /var/lib/asterisk/static_html The Asterisk GUI uses the web server built into Asterisk, so what you're attempting to do here isn't going to work. I suggest you follow the instructions at http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect111_tt1363.html. They may be a bit out of date (as the Asterisk GUI has changed quite a bit since we wrote the book), but it should help you get started. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
Since /etc/asterisk is empty, you have either relocated your conf files or put them in a database. Assuming neither, just create manager.conf in /etc/asterisk with this setup [general] Enabled = yes Port = 5038 Webenabled=yes Bindaddr = 1.2.3.4 [loginname] Secret=secret And restart asterisk _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 1:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] q: install asterisk + asteris-gui hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 /var/lib/asterisk/static_html now, i see the login box, but i dont have any credentials. tutorials are suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is empty... what am i supposed to do now? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: which Browser-GUI do u guys use?
On Wed, 8 Jul 2009, tom wrote: None. I'm a command line weenie. ) GUIs don't let you annotate your changes -- who did what (or what they thought they were doing), when, and why. ) GUIs don't support any sort of versioning. ) GUIs don't support any sort of configuration rollback. All of these are essential when something that used to work suddenly doesn't. (Sometimes, client's don't notice something isn't working for months -- way beyond my short term memory.) I'm sure I could come up with dozens more, these were just the first 3. (Probably not even the most important 3.) Oh. Here's 1 more -- GUIs impede truly understanding a system. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
/etc/manager.conf: [admin] secret = test read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config - doenst let me log in ;-( - i tried chown /static_http/config this is in my apache-logs: [Wed Jul 08 15:36:23 2009] [error] [client 66.134.175.166] File does not exist: /var/www/rawman, referer: http://123.456.789.999/pbx/config/home.html i did a symlink from /var/www/pbx to /var/lib/asterisk/static_html whats wromg here? thx again ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
Do you have the [general] section with enabled, webenabled, port and ipaddress? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] q: install asterisk + asteris-gui /etc/manager.conf: [admin] secret = test read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config - doenst let me log in ;-( - i tried chown /static_http/config this is in my apache-logs: [Wed Jul 08 15:36:23 2009] [error] [client 66.134.175.166] File does not exist: /var/www/rawman, referer: http://123.456.789.999/pbx/config/home.html i did a symlink from /var/www/pbx to /var/lib/asterisk/static_html whats wromg here? thx again ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
thx, but still struggeling: http://blabla:8088/asterisk/static/docs/index.html NO GO --- ; ; Asterisk Builtin mini-HTTP server ; ; ; Note about Asterisk documentation: ; If Asterisk was installed from a tarball, then the HTML documentation should ; be installed in the static-http/docs directory which is ; (/var/lib/asterisk/static-http/docs) on linux by default. If the Asterisk ; HTTP server is enabled in this file by setting the enabled, bindaddr, ; and bindport options, then you should be able to view the documentation ; remotely by browsing to: ; http://server_ip:bindport/static/docs/index.html ;[general] enabled=yes enablestatic=yes ; without this, you can only send AMI commands, not display ; html content bindaddr=0.0.0.0; address you want the Asterisk HTTP server to respond on bindport=8088 ; port you want the Asterisk HTTP server to respond on prefix=asterisk ; will form part of the URI, similar to a directory name - manager: [general] enabled=yes ; you may already have AMI enabled if you are using it for other things webenabled=yes ; this enables the interaction between the Asterisk web server and AMI [tom] ; you can name the user whatever you want secret = tom read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config --- i opened up port 8088 on the router AND on the iptables...unfortunately im outside of the network righ tnow how can i debug? any ideas? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
stupid me, i had a ; in front of the [general] line. thx so far im logged inand now? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
In http.conf make bindaddr be the address of your asterisk server. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 3:01 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] q: install asterisk + asteris-gui thx, but still struggeling: http://blabla:8088/asterisk/static/docs/index.html NO GO --- ; ; Asterisk Builtin mini-HTTP server ; ; ; Note about Asterisk documentation: ; If Asterisk was installed from a tarball, then the HTML documentation should ; be installed in the static-http/docs directory which is ; (/var/lib/asterisk/static-http/docs) on linux by default. If the Asterisk ; HTTP server is enabled in this file by setting the enabled, bindaddr, ; and bindport options, then you should be able to view the documentation ; remotely by browsing to: ; http://server_ip:bindport/static/docs/index.html ;[general] enabled=yes enablestatic=yes ; without this, you can only send AMI commands, not display ; html content bindaddr=0.0.0.0; address you want the Asterisk HTTP server to respond on bindport=8088 ; port you want the Asterisk HTTP server to respond on prefix=asterisk ; will form part of the URI, similar to a directory name - manager: [general] enabled=yes ; you may already have AMI enabled if you are using it for other things webenabled=yes ; this enables the interaction between the Asterisk web server and AMI [tom] ; you can name the user whatever you want secret = tom read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config --- i opened up port 8088 on the router AND on the iptables...unfortunately im outside of the network righ tnow how can i debug? any ideas? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream 2010 and blinky lights
Because DEVSTATE is for custom hints - and have you tried to set one every time a phone rings/is answered? This was thought about - but the logic in the dialplan would be a nightmare. Anyway, doing it the way I do it works for me (and others) as my dialplan contains nothing but 'include' and 'switch' statements now (so it reloads fast). Thanks for the reply though :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: 08 July 2009 09:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: aster...@dotr.com Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights On 8/7/09 8:52 PM, Andrew Thomas wrote: That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream 2010 and blinky lights Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. Why don't you just use func_devstate which was backported to 1.4? That way you can just set a DB variable on login/logout. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
yeah thx i did that. now if i log in ( :8088/asterisk/static/ajamdemo.html) , i see the Asterisk™ AJAM Demo. but thats it: i tries the urls givin by : http show status, but none of them gives me a real webinterface to administrate the whole asterisk etc i thought asterisk-gui gives me the ability to have a web-gui, right? thx tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
You're confusing the manager interface with the gui interface. The gui interface would be 8088/asterisk/static/config/index.html _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] q: install asterisk + asteris-gui yeah thx i did that. now if i log in ( :8088/asterisk/static/ajamdemo.html) , i see the AsteriskT AJAM Demo . but thats it: i tries the urls givin by : http show status, but none of them gives me a real webinterface to administrate the whole asterisk etc i thought asterisk-gui gives me the ability to have a web-gui, right? thx tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: q: install asterisk + asteris-gui: SOLVED
:8088/asterisk/static/config/index.html wes my missing link thx 2 all for ur help -- Forwarded message -- From: tom tomabr...@gmail.com Date: Wed, Jul 8, 2009 at 4:19 PM Subject: Re: [asterisk-users] q: install asterisk + asteris-gui To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com yeah thx i did that. now if i log in ( :8088/asterisk/static/ajamdemo.html) , i see the Asterisk™ AJAM Demo. but thats it: i tries the urls givin by : http show status, but none of them gives me a real webinterface to administrate the whole asterisk etc i thought asterisk-gui gives me the ability to have a web-gui, right? thx tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
thx again, one last question: as i mentioned, i used freepbx before. now i facing only the section: - users my goal right now is to use that asterisk instance just to have intenral extensions to talk to each other...whats the quickest setup here? i mean i dont need trunks, dialplans etc, right? i just need 5 internal extension, eg 1001-1005 thx u guys are great! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
If you're just going to use Asterisk as an internal system, you just need a
simple users.conf, sip.conf and about a 5 line dialplan.
Sip.conf
[general]
srvlookup=yes ;allows DNS lookups of server names
naxexpirey=180
defaultexpirey=160
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.23.95 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
limitonpeers=yes
notifyringing=yes
rtupdate=yes
artcachefriends=yes
notifyhold=yes
incominglimit=1
call-limit=3
[authentication]
[104]
type=peer
context=phones
host=dynamic
fromuser=104
secret=
canreinvite=yes
directrtpsetup=no
call-limit=3
nat=yes
qualify=yes
register=no
session-timers=accept
session-expires=60
session-minse=120
session-refresher=uac
register = 1001:x...@yourpbx.com/1001
defaultip=192.168.23.114
mailbox=1001
disallow=all
allow=ulaw,alaw
rinse and repeat for 1002-1005
users.conf
[general]
; Full name of a user
fullname = Unknown User
; Starting point of allocation of extensions
userbase = 1001
; Create voicemail mailbox and use use macro-stdexten
hasvoicemail = yes
; Set voicemail mailbox 1001 password to 1234
vmsecret = 1234
; Create SIP Peer
hassip = yes
; Create IAX friend
hasiax = no
; Create Agent friend
hasagent = no
; Create H.323 friend
;hash323 = yes
; Create manager entry
hasmanager = no
; Remaining options are not specific to users.conf entries but are general.
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
localextenlength = 4
[1001]
username=1001
transfer=yes
mailbox=1001
call-limit=3
fullname=user 1
registersip=no
host=dynamic
callgroup=1
context=default
cid_number=1001
hasvoicemail=yes
vmsecret=1234
email=us...@yourpbx.com
threewaycalling=yes
hasdirectory=no
callwaiting=yes
hasmanager=yes
managerread=system,call,log,verbose,command,agent,user,config
managerwrite=system,call,log,verbose,command,agent,user,config
hasagent=yes
hassip=yes
hasiax=no
secret=
nat=yes
canreinvite=no
dtmfmode=rfc2833
insecure=no
pickupgroup=1
macaddress=
autoprov=yes
label=100
linenumber=1
disallow=all
allow=ulaw,gsm
repeat for 1002-1005
extensions.conf
[default]
Exten = s,1,answer
Exten = s,n,hangup
Exten = _1XXX,1,Dial(SIP/${EXTEN},60.m)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom
Sent: Wednesday, July 08, 2009 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] q: install asterisk + asteris-gui
thx again,
one last question: as i mentioned, i used freepbx before. now i facing only
the section:
- users
my goal right now is to use that asterisk instance just to have intenral
extensions to talk to each other...whats the quickest setup here? i mean i
dont need trunks, dialplans etc, right?
i just need 5 internal extension, eg 1001-1005
thx
u guys are great!
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[asterisk-users] Queue autopause
Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk 1.4.21.2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
thx danny, (sorry, bad day today) one more question: deviceandusers i had this distinction with freepbx, though i dont know whether this is a freepbx-thing or an asterisk-setting... thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: install asterisk + asteris-gui
That's a new one on me, but check out this link http://forums.digium.com/viewtopic.php?t=3689 http://forums.digium.com/viewtopic.php?t=3689highlight=sid=acbc25fd45bae1 ecc42b0d7ca66fe88c highlight=sid=acbc25fd45bae1ecc42b0d7ca66fe88c As I read it, you want to be able to dial 1001 and get the user of 1001 wherever he or she is? If so, FOLLOWME is supposedly the way to do that. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] q: install asterisk + asteris-gui thx danny, (sorry, bad day today) one more question: deviceandusers i had this distinction with freepbx, though i dont know whether this is a freepbx-thing or an asterisk-setting... thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
Barry D. Hassler wrote: Well, Teliax says they have no access to the PSTN's database, but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). As I understand it, if they got a document signed by their origination provider granting them authorization to do CNAM hosting on their own numbers, they could then hire someone such as Verisign to host their CNAM records in the so-called PSTN database. They'd even profit from this assuming they have enough subscribers. There are probably several reasons for why they don't do this, possibly starting with administrative overhead and/or their provider is not willing to relinquish control of the records. If someone has experience with this, feel free to correct me. However, this is my understanding from my previous experience with looking up Caller Name information via CNAM/LIDB/SS7. Regards, -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250 483-0386 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using a mobile phone via USB as an extension
Thank you for the info Does anyone know if the cdc-modem interface which is available on mobile phones can actually potentially be used to initiate, or register for receiving a voice call? If so, I suppose USB 3G dongles could even be used as a voip-air interface! Would be interesting to find specs for these. Administrator TOOTAI wrote: Carlos Ruiz Diaz a écrit : Check chan_mobile. [...] Or use GSM gateway On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote: I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is the same as a land line. In other countries, it is very different. I know of a mobile operator who bundle lots of free on-network minutes with SIM cards. I wonder if it is possible to forward the call via a mobile phone tethered to an asterisk server through USB? Has anyone tried tethering a mobile phone to an asterisk server and configuring it as an asterisk extension so they can use free or cheap on-network minutes for the mobile leg of the call? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue autopause
Christian Gansberger escribió: Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk 1.4.21.2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why would you want to do that? The purpose of the autopause is to discard the absent agent that is not responding to the calls to not try it anymore until it gets unpaused by a supervisor or someone else, and therefore the pause is made to all queues the agent is member of. Why pause it on only one queue, letting it ring on other queues? Aside from the purpose you have on this, I think you would need to modify the app_queue.c code to make the parameter configurable inside each queue definition and not on the general section of queues.conf. Then you would need to modify the logic to handle the autopause configured for each queue. This is a general idea as I didn't take a deep look of app_queue.c to see how it works exactly. Any other solution without changing asterisk code would imply a external application that monitors the queues and makes the custom autopause you need. Just my two cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g.722 + loudness
What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com): Kevin P. Fleming wrote: Hose wrote: I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). It is not in the 1.6.0.10 Changelog nor the 1.6.1.1 Changelog. It is, however, in the 1.6.0.11-rc1 Changelog and the 1.6.2.0-beta3 Changelog. -Dave Interesting - the one time I didn't bother to upgrade it gets listed as fixed in the changelog. However I did upgrade just now after a break in calls to .10. Unfortunately it's still the same, but might the change not have been implemented in .10 according to Dave's notes? The workaround at the moment is just to use ulaw and that works (though obviously without the wonders that is g.722). If it's in the pipline for .11, I can just hang out until then as I'd prefer not to run an -rc version. hose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
It does which is why it was not included in a release code set. The patch could be changed to do an OR type compare for the bridge class. I have changed my implementation to use only user events for everything that I now need so I did not pursue this patch. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Benny Amorsen benny+use...@amorsen.dk Date: Tue, 07 Jul 2009 17:32:11 +0200 To: Jim Dickenson dicken...@cfmc.com Cc: Asterisk User MailList asterisk-users@lists.digium.com Subject: Re: What is the best way to share extension state Jim Dickenson dicken...@cfmc.com writes: http://bugs.digium.com/view.php?id=14595 has a patch to add a new class, bridge, so you get less events in AMI. This is for 1.6.0.x. It will give you an idea of what needs to be changed in order to make the call class of messages more granular. It's nice to see that work is done to make it more granular. However, doesn't that break backwards compatibility, in the people who request call now don't get bridge events? The challenge is that eventually every single manager_event will have its own type... Anyway, that's a worry for another time. It's really neat! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g.722 + loudness
Hose wrote: What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com): Kevin P. Fleming wrote: Hose wrote: I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). It is not in the 1.6.0.10 Changelog nor the 1.6.1.1 Changelog. It is, however, in the 1.6.0.11-rc1 Changelog and the 1.6.2.0-beta3 Changelog. -Dave Interesting - the one time I didn't bother to upgrade it gets listed as fixed in the changelog. However I did upgrade just now after a break in calls to .10. Unfortunately it's still the same, but might the change not have been implemented in .10 according to Dave's notes? The workaround at the moment is just to use ulaw and that works (though obviously without the wonders that is g.722). If it's in the pipline for .11, I can just hang out until then as I'd prefer not to run an -rc version. Even when a loud voice drives the G.722 into trouble, it still sounds better than crappy u-Law. :-) Only a few lines were changed in the G.722 codec files. You could easily replicate those changes in an older version of Asterisk, if you need to. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] q: sip registration fails...
[Jul 8 21:23:49] WARNING[4358]: chan_sip.c:10458 check_auth: username mismatch, have 6001, digest has 1160 [Jul 8 21:23:49] NOTICE[4358]: chan_sip.c:18529 handle_request_register: Registration from 'sip:6...@192.168.1.4 sip%3a6...@192.168.1.4' failed for '192.168.1.3' - Username/auth name mismatch sip.conf [6001] user=6001 type = friend secret=6001 host = dynamic callerid = 6005 context = from-sip-internal allow=all i dont find anything with 1160, where does that come from? thx 4 looking ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] q: am i mixing somethign up?
hi, checking my freshly installed astersik-gui, i can see a menu entry called Users. clicking on that one gives me the pages labeled (on orange) User Extensions on PBX. if i do make an entry here, it ends up in the user.conf. file. so i created a new entry in the sip.conf, reloaded asterisk cant see it anywhere in the guimaybe im just confused here right now thx 4 clarification ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial stops trying after ~30s regardless
Hi,
My Dial() is set to the following, but always stops about 30 seconds into
the call even when I set it to try for 60 seconds.
exten = dialnumber,1,Dial(${DialInfo},60)
I am running on 1.6.1-r199820.
Is there some other setting that is overriding mine? Or an issue with this
release? Thanks for the help.
JR
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Re: [asterisk-users] calculate data traffic
On 9/7/09 12:11 AM, jonas kellens wrote: To calculate the monthly data traffic that is generated by VoIP-calls, is it as simpel as 80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month = 585.9375 MB traffic / month http://www.asteriskguru.com/tools/bandwidth_calculator.php -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] q: am i mixing somethign up?
On 9/7/09 1:39 PM, tom wrote: hi, checking my freshly installed astersik-gui, i can see a menu entry called Users. clicking on that one gives me the pages labeled (on orange) User Extensions on PBX. if i do make an entry here, it ends up in the user.conf. file. so i created a new entry in the sip.conf, reloaded asterisk cant see it anywhere in the guimaybe im just confused here right now You might be best asking in the Asterisk-GUI mailing list. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial stops trying after ~30s regardless
On 9/7/09 2:06 PM, John Regal wrote:
Hi,
My Dial() is set to the following, but always stops about 30 seconds
into the call even when I set it to try for 60 seconds.
exten = dialnumber,1,Dial(${DialInfo},60)
I am running on 1.6.1-r199820.
Is there some other setting that is overriding mine? Or an issue with
this release? Thanks for the help.
Obviously you're not typing dialnumber on your phone keypad, so likely
you are initiating the call from a call file or the manager.
Check to see what you have set for the timeout value in whatever method
you use to make the call.
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] Dial stops trying after ~30s regardless
Hi - yes, you are correct in that I am using AMI. I thought I could override
inline in the dialplan. I will modify the AMI call. Thanks for the quick
response - truly appreciated.
john
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Wednesday, July 08, 2009 10:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial stops trying after ~30s regardless
On 9/7/09 2:06 PM, John Regal wrote:
Hi,
My Dial() is set to the following, but always stops about 30 seconds
into the call even when I set it to try for 60 seconds.
exten = dialnumber,1,Dial(${DialInfo},60)
I am running on 1.6.1-r199820.
Is there some other setting that is overriding mine? Or an issue with
this release? Thanks for the help.
Obviously you're not typing dialnumber on your phone keypad, so likely
you are initiating the call from a call file or the manager.
Check to see what you have set for the timeout value in whatever method
you use to make the call.
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] Anonymous Connection form IP to use specific Context
Hi All, I never saw a reply to this question. Is anyone able to assist? Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Friday, 19 June 2009 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Anonymous Connection form IP to use specific Context Hi All, How can I force an anonymous SIP connection from a certain IP address to use a specific context rather than the default one defined in sip.conf. I am using Asterisk 1.6.0.9 Regards David Klaverstyn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
