[asterisk-users] FW: Change the FROM filed username and From
Hello Guys, Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK-966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. * *From: asterisksip:aster...@ip1:5065;tag=as0cae0b** see the last part this is what that i want to change here in from it should be some CLI thanks Masood ___ -- 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] Crosstalk - Is there a debug option for logging this?
JT djklut...@gmail.com writes: I'm struggling with an intermittent crosstalk issue resulting in a caller's audio being broadcasted to other calls (only one way as they are unable to hear the others listening in). This may be a long shot... I have experienced this when two SIP phones had the same IP address (a bug by itself of course). Now, obviously the SIP phone should not just play any random audio that someone throws at it, but apparently life is not so simple. /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
[asterisk-users] ChanIsAvail querry
Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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] Experience with LLDP
Warren Selby wcse...@selbytech.com writes: I believe I spoke with Aastra and Snom at the Astricon tradeshow and they said they support it on their newer models as well. For Snom the enhancement request is SCPP-227, but I don't believe it has been implemented. I can't find it in any release notes at least. The general public can't track SCPP's, which is a bit inconvenient. /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
[asterisk-users] DGP 301hard phone incomming problem.
Dear all, i am using DGP 301 hard phone with my asterisk server. 1 : real time support is enabled ...all sip_buddies are stored in mysql database... 2: when i register my phone for first time it works fine.receives 2 ,3 calls then no call received hangup cause is congestioni don't know why. 3: when i unregister or shutdown my dgp 301 hard phone .it still visible as registered in real time database. but not visible as online in asterisk cache... i have spent lot of toime to point out the probelm but cant. can any body help me in this regrad. -- Best Regards Yawar Hadi Noshahi Consultant/Software Engineer NGI Islamabad MS Computer Science Linkoping University Sweden +46700-445479 ___ -- 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] ChanIsAvail querry
On 14:59, Wed 25 Nov 09, ABBAS SHAKEEL wrote: Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest As the documentation will tell you: This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists So check the contents of that variable after running ChanIsAvail() -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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
[asterisk-users] How many lines do you use.
Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... Julian ___ -- 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] ChanIsAvail querry
What version of Asterisk are you using? I think this might be related to an issue that was resolved in version 1.4.27 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html - look in the list of Closed Items, second one down. https://issues.asterisk.org/view.php?id=14426 - link to the issue Hope that helps. Dan Journo From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: 25 November 2009 09:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanIsAvail querry Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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] 1950's UK rotary dial phone
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote: Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true Should not be needed. This parameter means that your Asterisk system dials with pulses rather than tones (prefixing 't' rather than 'p' to the dial string sent to DAHDI). I suppose this is not really what you're after. or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on the hook. However, using the rotary dialer does not work (works fine plugged into my phone line). I have read about the possibilty that the pulse settings may need adjusting in kernel.h in the dahdi driver but I have no idea what to set them to. I have tried tweeking them to various extents but I've not been able to bring it to life yet. Does anyone have any experience getting this to work? Does anyone know the specs for UK pulse dial? How long should the pulses be and what is the gap between them? I wonder if anybody wants to follow up on http://bugs.debian.org/546329 (formly http://bugs.debian.org/399772 ) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] How many lines do you use.
On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Please some enlightment on ENUM !!
SIP schrieb: Yes... you would have to register (and possibly pay for, dependent on the ENUM registrar) each individual number. The idea behind ENUM is that it's an E164 number that is already yours that maps to whatever you want it to map to (email, SIP, etc). The key point here is that you already own the E164 number. If you do, then you could register them all at e164.org for free. If you don't own the individual numbers, you shouldn't be allowed to register them as your own. That sort of breaks the ENUM concept of a number you take with you as a personal identifier. N. ___ -- 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 Hi N. ! Thanks for your answer. Either I don't understand what you want to tell me or this thread slowly drifts away from my original question. My original question was: If you own a telephone number which connects to your company and you have a PBX (like asterisk) and some extesniosn behind that, how/where do you enum-register each extension so that each extension can be reached from outside by a SIP uri? Meanwhile I managed to speak to a technician at my-enum.at, which is my registrar at e164.arpa. He *comfirmed* my original assumption: If you have a telephone number and want to paticipate in enum, you have to register that number at - for example - e164.arpa. If you operate extensions behind that number and you want them to be reachable too, you have to run your own DNS server and register this server at e164.arpa. This server is naturally under your responsibility and you manage all your extension yourself. It is works exactly like any other DNS resolution. Norbert ___ -- 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] ChanIsAvail querry
Thanks Michiel and Dan @ Michiel i have checked the variables but they dont contain any value. @Dan I am using 1.6.1.2 May be some issue with it ... In the mean while let me test with an older version of asterisk On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo d...@keshercommunications.comwrote: What version of Asterisk are you using? I think this might be related to an issue that was resolved in version 1.4.27 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html- look in the list of Closed Items, second one down. https://issues.asterisk.org/view.php?id=14426 – link to the issue Hope that helps. Dan Journo *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL *Sent:* 25 November 2009 09:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanIsAvail querry Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] Route Non-Call Data to Agent Through Queue
Yes why not? when the agent is connected it can read the variables on the calling channel what would you like to build with that? :) l. 2009/11/24 Shaun Clark shaun_cl...@hotmail.com Hello, I was wondering if their is a way to use the Asterisk ACD to initiate a call that will route variables through the ACD, which can then be read at the other end by an application. The idea here is instead of terminating a call to an agent I would be terminating some variables/text data. Thanks! Shaun -- Loway - home of QueueMetrics - http://queuemetrics.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] 1950's UK rotary dial phone
On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote: On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote: Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true Should not be needed. This parameter means that your Asterisk system dials with pulses rather than tones (prefixing 't' rather than 'p' to the dial string sent to DAHDI). I suppose this is not really what you're after. or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on the hook. However, using the rotary dialer does not work (works fine plugged into my phone line). I have read about the possibilty that the pulse settings may need adjusting in kernel.h in the dahdi driver but I have no idea what to set them to. I have tried tweeking them to various extents but I've not been able to bring it to life yet. Does anyone have any experience getting this to work? Does anyone know the specs for UK pulse dial? How long should the pulses be and what is the gap between them? I wonder if anybody wants to follow up on http://bugs.debian.org/546329 (formly http://bugs.debian.org/399772 ) Thanks Tzafrir, that was it. I have changed the fxs.debounce and recompiled. My rotary dialer lives again! Thanks again, Mike. signature.asc Description: Digital signature ___ -- 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] ChanIsAvail querry
Dan I have reverted to 1.4.27 but got no success. Same behaviour Do anyone has any success with it ? On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Thanks Michiel and Dan @ Michiel i have checked the variables but they dont contain any value. @Dan I am using 1.6.1.2 May be some issue with it ... In the mean while let me test with an older version of asterisk On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo d...@keshercommunications.comwrote: What version of Asterisk are you using? I think this might be related to an issue that was resolved in version 1.4.27 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html- look in the list of Closed Items, second one down. https://issues.asterisk.org/view.php?id=14426 – link to the issue Hope that helps. Dan Journo *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL *Sent:* 25 November 2009 09:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanIsAvail querry Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- 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] ChanIsAvail querry
On 16:54, Wed 25 Nov 09, ABBAS SHAKEEL wrote:
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
This ael snippet is working great for me on current -trunk.
I have been using this for some time now, it's from before 1.6 got
branched so it should work there as well I think.
Verbose(1,Routing call from ${CALLERID(num)} (${CALLERID(name)}) to
${EXTEN} on channel ${CHANNEL});
ChanIsAvail(Skinny/6000Skinny/6002SIP/michiele71,a);
if ( x${AVAILORIGCHAN} != x ) {
Verbose(1,Calling available channels: ${AVAILORIGCHAN});
Dial(${AVAILORIGCHAN},45,htxk);
}
On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Michiel and Dan
@ Michiel i have checked the variables but they dont contain any value.
@Dan I am using 1.6.1.2 May be some issue with it ... In the mean while
let me test with an older version of asterisk
On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo
d...@keshercommunications.comwrote:
What version of Asterisk are you using?
I think this might be related to an issue that was resolved in version
1.4.27
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html-
look in the list of Closed Items, second one down.
https://issues.asterisk.org/view.php?id=14426 ? link to the issue
Hope that helps.
Dan Journo
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL
*Sent:* 25 November 2009 09:59
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanIsAvail querry
Hello
We need to know if a channel is not in use and can be used to dial a
number etc..
I have tried the ChanIsAvail function with different parameters.
ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc
no matter the channel is busy or not it always return 0 .
Please suggest
FYI
ChanIsAvail(Technology/resource[Technology2/resource2...][,options]):
This application will check to see if any of the specified channels are
available.
Options:
a - Check for all available channels, not only the first one.
s - Consider the channel unavailable if the channel is in use at all.
t - Simply checks if specified channels exist in the channel list
(implies option s).
This application sets the following channel variable upon completion:
AVAILCHAN - the name of the available channel, if one exists
AVAILORIGCHAN - the canonical channel name that was used to create the
channel
AVAILSTATUS - the status code for the available channel
--
Best Regards
Shakeel Abbas
___
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--
Best Regards
Shakeel Abbas
--
Best Regards
Shakeel Abbas
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--
Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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[asterisk-users] asterisk + res_config_ldap = asterisk.core
Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users' declarations in sip.conf) everything works fine. Software info: amd64 FreeBSD 8.0-RC1. asterisk16-1.6.0.15, latest from ports. OpenLDAP. Configs and logs: dummyuzer attributes in LDAP database: objectClass AsteriskSIPUser uid dummyuzer AstAccountCallerID 2001 AstAccountContext domain AstAccountDefaultUser dummyuzer AstAccountExpirationTimestamp 1259080333 (and it changes, so this part is ok) AstAccountHost dynamic AstAccountIPAddress 10.3.8.104 (and it changes from default random address, so it's ok too) AstAccountLastQualifyMilliseconds 0 AstAccountNAT no AstAccountPort 5060 AstAccountQualify no AstAccountType friend sip.conf: [general] allow=all bindaddr=0.0.0.0 realm=gw.domain.com extconfig.conf: [settings] sipusers = ldap,ou=users,o=domain,sip sippeers = ldap,ou=users,o=domain,sip extensions = ldap,o=domain,extensions res_ldap.conf: [_general] url=ldap://xx.xx.xx.xx:389 protocol=3 basedn=ou=users,o=domain user=uid=asterisk,ou=virtual,o=domain pass=X [config] additionalFilter=(objectClass=AstConfig) filename = AstConfigFilename category = AstConfigCategory variable_name = AstConfigVariableName variable_value = AstConfigVariableValue cat_metric = AstConfigCategoryMetric commented = AstConfigCommented [extensions] context = AstContext exten = AstExtension priority = AstPriority app = AstApplication appdata = AstApplicationData additionalFilter=(objectClass=AsteriskExtension) [sip] name = uid amaflags = AstAccountAMAFlags callgroup = AstAccountCallGroup callerid = AstAccountCallerID canreinvite = AstAccountCanReinvite context = AstAccountContext dtmfmode = AstAccountDTMFMode fromuser = AstAccountFromUser fromdomain = AstAccountFromDomain fullcontact = AstAccountFullContact host = AstAccountHost ipaddr = AstAccountIPAddress insecure = AstAccountInsecure mailbox = AstAccountMailbox md5secret = AstAccountRealmedPassword nat = AstAccountNAT deny = AstAccountDeny permit = AstAccountPermit pickupgroup = AstAccountPickupGroup port = AstAccountPort qualify = AstAccountQualify restrictcid = AstAccountRestrictCID rtptimeout = AstAccountRTPTimeout rtpholdtimeout = AstAccountRTPHoldTimeout type = AstAccountType disallow = AstAccountDisallowedCodec allow = AstAccountAllowedCodec MusicOnHold = AstAccountMusicOnHold regseconds = AstAccountExpirationTimestamp regcontext = AstAccountRegistrationContext regexten = AstAccountRegistrationExten CanCallForward = AstAccountCanCallForward defaultuser = AstAccountDefaultUser regserver = AstAccountRegistrationServer lastms = AstAccountLastQualifyMilliseconds additionalFilter = (objectClass=AsteriskSIPUser) modules.conf: [modules] autoload=yes preload = func_strings.so noload = pbx_gtkconsole.so load = res_musiconhold.so noload = chan_alsa.so noload = chan_oss.so noload = chan_console.so noload = chan_ooh323.so noload = chan_skinny.so noload = chan_h323.so noload = cdr_odbc.so noload = res_config_sqlite.so noload = res_config_odbc.so noload = res_jabber.so noload = cdr_pgsql.so noload = cdr_sqlite.so noload = res_config_pgsql.so noload = cdr_sqlite3_custom.so noload = chan_gtalk.so noload = chan_jingle.so noload = res_snmp.so noload = res_odbc.so noload = cdr_adaptive_odbc.so noload = res_smdi.so noload = res_phoneprov.so noload = res_musiconhold.so noload = func_odbc.so noload = chan_unistim.so noload = app_minivm.so noload = chan_agent.so noload = chan_iax2.so noload = app_voicemail.so noload = chan_mgcp.so noload = pbx_ael.so noload = res_config_curl.so noload = cdr_custom.so noload = cdr_radius.so part of /var/log/asterisk/full (it always same): [Nov 25 14:39:45] NOTICE[87569] loader.c: 1 modules will be loaded. [Nov 25 14:39:45] NOTICE[87569] cdr.c: CDR simple logging enabled. [Nov 25 14:39:45] NOTICE[87569] loader.c: 139 modules will be loaded. [Nov 25 14:39:45] NOTICE[87569] config.c: Registered Config Engine ldap [Nov 25 14:39:45] WARNING[87569] translate.c: plc_samples 160 format f [Nov 25 14:39:45] VERBOSE[87569] logger.c: SIP channel loading... part of slapd.log: Nov 24 15:12:55
Re: [asterisk-users] How many lines do you use.
I use two 'lines' though 'Line appearances' would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, November 25, 2009 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How many lines do you use. Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... Julian ___ -- 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] [SOLVED] Cianet channel bank with noise and echo
snip Thats probably it You're relying on Asterisks software echo canceling I have seen mixed results. Have you tried adjusting gains? I'd do the following 1. Turn off echo canceler (makes it more obvious whilst you're trying to remove it) 2. Turn down both gains 3. listening' inside your network (i.e. listening to audio coming to your network from the PSTN), adjust the gain upwards until it sounds suitable 4. 'listening' outside of your network (i.e. listening to audio coming from your network to PSTN) do the same. 5. Test for echo. Adjust the gains down for the sound you hear back (i.e. if you hear person inside your network echoing, adjust the gain in '4'). 6. Try and get it as close to echo free as you can using this method 7. Enable any echo canceling you can find to 'tidy up' the leftover echo, try various ones if you need to. voip-info.org/wiki may help 8. Buy hardware echo cancelers next time ;) My echo problem was ajust gain, and enable echocanceller=mg2,1 for dahdi channels. Thanks for the help and troubleshooting guide. [ ] 's ___ -- 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] ChanIsAvail querry
I wonder if this is related to my problem where the channel returns with a status of BUSY even if it is on hook -- this is a dahdi channel. ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Dan I have reverted to 1.4.27 but got no success. Same behaviour Do anyone has any success with it ? On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Thanks Michiel and Dan @ Michiel i have checked the variables but they dont contain any value. @Dan I am using 1.6.1.2 May be some issue with it ... In the mean while let me test with an older version of asterisk On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo d...@keshercommunications.comwrote: What version of Asterisk are you using? I think this might be related to an issue that was resolved in version 1.4.27 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html- look in the list of Closed Items, second one down. https://issues.asterisk.org/view.php?id=14426 – link to the issue Hope that helps. Dan Journo *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL *Sent:* 25 November 2009 09:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanIsAvail querry Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas Alternatives: ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] hardware echo cancellation
If I get an echo cancellation module for my Digium TE121 card, will I need to do any adjustments/configuration in Asterisk? You should probably still set the gain using rxgain and txgain. IME, it's much easier setting gains on a PRI than it is on a POTS line, though. I've worked with a couple of PRIs that need no adjustments at all. Is the hardware better than the software version? The hardware version is the same algorithm as the HPEC echo canceler. It's quite a bit better than the MG2 algorithm that comes free with asterisk and maybe slightly better than OSLEC. The convergence time of the hardware algorithm is pretty fast (time it takes for the EC to effectively get rid of echo on a call). FYI: If you're considering running the software-based HPEC for all channels on a T1/E1, you should use a reasonably fast machine, as it uses quite a bit of CPU. That's one big reason to get the hardware module. - Noah TIA! ___ -- 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] Connect two Asterisk Server in IAX ?
I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. To clarify and expand on Aggio's response. You either need to have a peer and user on both machines, or you can set it up as type=friend, which is the peer and user combined. - Noah ___ -- 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] How many lines do you use.
On Wed, Nov 25, 2009 at 5:18 AM, Julian Lyndon-Smith aster...@dotr.com wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... Cisco describes those as 'buttons' as you can map them to features as you wish, such as speed dial that the one poster mentioned. I have one set up as a test/debug phone, where I have five buttons mapped to different asterisk systems where the phone registers. This lets me dial direct and do better testing specific to the roll of a given server. My polycom phones only have two lines or buttons or whatever you want to call them, and are less useful for that specific purpose. I agree that if you only have the phone registering with a single server, six simultaneous lines seems like overkill. ___ -- 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] How many lines do you use.
I use two ‘lines’ though ‘Line appearances’ would be a better term, though still confusing in my book. I have five line appearances on the Snom190 on my desk. I regularly use two line appearances, and on occasion, I have used three to juggle back and forth between calls. I would guess that a busy receptionist might have to use up to 6 line appearances all at once, but I can't imagine one person being able to use much more than that. I think most people get those sidecar units to do speed dials or to monitor other extensions. It's an interesting question, though. I regularly recommend Polycom 550s to my clients, but I would guess that 450s or even 335s would be just fine for most people. - Noah ___ -- 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] Questions about Voicemail
On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote: Regarding the email to multiple receipients, it is available on an ad-hoc basis from the phone? IE; call into the voicemail system, enter x digit to send a voicemail to multiple users, record the message, then enter the destination mailboxes, separated by # sign... You can enable an option in the voicemail that allows the prompt: 'To send a message to another user'... sendvoicemail=yes ; Allow the user to compose and send a voicemail ; while inside VoiceMailMain() [option 5 from ; mailbox's advanced menu]. If set to 'no', ; option 5 will not be listed. This would enable the option from within the vm app, but you want to do a dynamic list of mailboxes to deliver to, so by the time we get here, I think it's going to be to late to to anything useful (since we already called the voicemail app.) You could write some dialplan magic with a while loop, so that the user can dial a specific extn (maybe call it 'group message') and then it will prompt for a mailbox number, followed by #, or just # to end. Then it could build this list of mailboxes as a variable before calling the voicemail app. I can attempt to build an example if you are interested. Rob ___ -- 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] How many lines do you use.
I use Polycom 501's with 2 LA's for production and 1 for testing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller Sent: Wednesday, November 25, 2009 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How many lines do you use. I use two 'lines' though 'Line appearances' would be a better term, though still confusing in my book. I have five line appearances on the Snom190 on my desk. I regularly use two line appearances, and on occasion, I have used three to juggle back and forth between calls. I would guess that a busy receptionist might have to use up to 6 line appearances all at once, but I can't imagine one person being able to use much more than that. I think most people get those sidecar units to do speed dials or to monitor other extensions. It's an interesting question, though. I regularly recommend Polycom 550s to my clients, but I would guess that 450s or even 335s would be just fine for most people. - Noah ___ -- 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
[asterisk-users] office / homeuser
hi, we are running a switchvox system, and i would like to know what the practice is for users who are working party in the main office and on some other days with their laptops either from home of on the road... right now i told them to unplugg the hardphone, coz having a softphone and the hardphone plugged at the same time doesnt work so: - two extensions for each location per user? - one extension which leads into a ringroup? so, what the best plan here? 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] How many lines do you use.
On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote: I use two ‘lines’ though ‘Line appearances’ would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) If you enable two line keys with the same user/pass then the phone will automatically put a second call/call waiting onto the second line key (assuming you have call waiting enabled.) But personally I preferred the way it presented the second call before, on a single line, and found the way it displays it with two lines a bit confusing. (I can't remember exactly why now, something like it would flash the second line icon but not show you the call information until that key was pressed, or you scrolled to it.) I could see users not getting on with this, so I didn't configure it like that. The rest can be used for speed dials, but these were of limited use to me since for some reason, although the line keys can be provisioned remotely over TFTP, the speed dials cannot. It's okay for personal use though. Personally moved off my 7960 in favour of the SNOM 370 as this supports far more features than the Cisco SIP image, which is only really a piece of migration fluff to enable Cisco to migrate customers away from competitors SIP systems onto Call Manager with the dual-boot/application loader. The SNOM perhaps doesn't look as fancy as the Cisco handset, but it wins hands-down on SIP features. (the remote provisioning system was a little complicated to set up, but once set up it's okay.) It's a shame since the Cisco is a very capable (and expensive) handset, just let down by no development in the software other than small bug fixes for many years. ___ -- 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] Questions about Voicemail
Rob; That would be great. You could send directly to me @ dovey.for...@idt.net or respond to this list. I appreciate it! --Dovey -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: Wednesday, November 25, 2009 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about Voicemail On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote: Regarding the email to multiple receipients, it is available on an ad-hoc basis from the phone? IE; call into the voicemail system, enter x digit to send a voicemail to multiple users, record the message, then enter the destination mailboxes, separated by # sign... You can enable an option in the voicemail that allows the prompt: 'To send a message to another user'... sendvoicemail=yes ; Allow the user to compose and send a voicemail ; while inside VoiceMailMain() [option 5 from ; mailbox's advanced menu]. If set to 'no', ; option 5 will not be listed. This would enable the option from within the vm app, but you want to do a dynamic list of mailboxes to deliver to, so by the time we get here, I think it's going to be to late to to anything useful (since we already called the voicemail app.) You could write some dialplan magic with a while loop, so that the user can dial a specific extn (maybe call it 'group message') and then it will prompt for a mailbox number, followed by #, or just # to end. Then it could build this list of mailboxes as a variable before calling the voicemail app. I can attempt to build an example if you are interested. Rob ___ -- 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] office / homeuser
I setup another extension for the softphone and enable followme on their main extension to ring both. For example 8678 is the main and 38678 is their softphone. For users with more phones I just keep going up 48678. This makes it fairly seamless to the end user and easy enough to remember when looking at the extension list. Ryan On Wed, Nov 25, 2009 at 11:58 AM, tom tomabr...@gmail.com wrote: hi, we are running a switchvox system, and i would like to know what the practice is for users who are working party in the main office and on some other days with their laptops either from home of on the road... right now i told them to unplugg the hardphone, coz having a softphone and the hardphone plugged at the same time doesnt work so: - two extensions for each location per user? - one extension which leads into a ringroup? so, what the best plan here? 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] ChanIsAvail querry
Thanks Michiel Covici @ Michiel i will try the script @Covice yes it is a DAHDI channel On Wed, Nov 25, 2009 at 8:12 PM, cov...@ccs.covici.com wrote: I wonder if this is related to my problem where the channel returns with a status of BUSY even if it is on hook -- this is a dahdi channel. ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Dan I have reverted to 1.4.27 but got no success. Same behaviour Do anyone has any success with it ? On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Thanks Michiel and Dan @ Michiel i have checked the variables but they dont contain any value. @Dan I am using 1.6.1.2 May be some issue with it ... In the mean while let me test with an older version of asterisk On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo d...@keshercommunications.comwrote: What version of Asterisk are you using? I think this might be related to an issue that was resolved in version 1.4.27 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html-look in the list of Closed Items, second one down. https://issues.asterisk.org/view.php?id=14426 – link to the issue Hope that helps. Dan Journo *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL *Sent:* 25 November 2009 09:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanIsAvail querry Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI ChanIsAvail(Technology/resource[Technology2/resource2...][,options]): This application will check to see if any of the specified channels are available. Options: a - Check for all available channels, not only the first one. s - Consider the channel unavailable if the channel is in use at all. t - Simply checks if specified channels exist in the channel list (implies option s). This application sets the following channel variable upon completion: AVAILCHAN - the name of the available channel, if one exists AVAILORIGCHAN - the canonical channel name that was used to create the channel AVAILSTATUS - the status code for the available channel -- Best Regards Shakeel Abbas ___ -- 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 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas Alternatives: ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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 -- Best Regards Shakeel Abbas ___ -- 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] How many lines do you use.
At 02:18 AM 11/25/2009, you wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) I've fought with the same question. When I first set it up, every phone had 4 sip lines so I could tell where the call came from. As my skills and understanding got better I changed them all back to one and use the dialplan the modify the callerid Name to tell me where the calls came from. It seems to work much better this way and it's a lot easier to keep track of what's going on. Also, the phone understands how to deal with 9 calls at once if you just let it be and if you start assigning a line to each button it doesn't seem to handle that kind of thing as well. Aastra 480i-CT. Ira ___ -- 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] How many lines do you use.
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) /snip Are you sure about this? I believe the 79xx series on 8x SIP firmware loads does BLF with SIP/TCP, just not SIP/UDP. -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] How many lines do you use.
snip On Wed, Nov 25, 2009 at 8:18 AM, Julian Lyndon-Smith aster...@dotr.comwrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) /snip Polycom 501 with 3 lines: 1 - Office Line 2 - Personal Line 3 - Tests ___ -- 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] office / homeuser
hi ryan, thx for ur suggestions. so , if i would go that route, that would mean i end up with n-extensions per user based on n-locations. questions: - how would i set the 'main' extension, so that other people see only one extension in the phonebook / have to remember? - is the caller-id when that person initiates a call as well the main-ext or do is this a setting somewhere? thx again regs tom On Wed, Nov 25, 2009 at 12:16 PM, Ryan Wagoner rswago...@gmail.com wrote: I setup another extension for the softphone and enable followme on their main extension to ring both. For example 8678 is the main and 38678 is their softphone. For users with more phones I just keep going up 48678. This makes it fairly seamless to the end user and easy enough to remember when looking at the extension list. Ryan On Wed, Nov 25, 2009 at 11:58 AM, tom tomabr...@gmail.com wrote: hi, we are running a switchvox system, and i would like to know what the practice is for users who are working party in the main office and on some other days with their laptops either from home of on the road... right now i told them to unplugg the hardphone, coz having a softphone and the hardphone plugged at the same time doesnt work so: - two extensions for each location per user? - one extension which leads into a ringroup? so, what the best plan here? 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 ___ -- 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] Channel Variable
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel variable seems to be wrong.
I have been using Zoiper as my IAX client and Asterisk 1.6.2.0-rc6.
For the sake of debugging I have Verbose(1,Outgoing Call Handler
${CUT(CHANNEL,-,1)}) in the (internal - not default) dial plan.
Most of the time the channel variable is IAX2/10007 which is the desired
behaviour (with 10007 being the IAX username) but some of the time
IAX2/192.168.1.111:4569 is shown instead.
I would like to know why this is happening and if there is anything that can be
done to make it show the IAX2/10007 form every time?
I realise that I could use ${CDR(accountcode)} instead, and as it happens this
returns the correct account code value in both cases. However, I wanted to be
able to do this on a per-channel basis and multiple channels currently share a
common accountcode.
Any ideas what's going on here, is there something obvious I'm missing?
Thanks in advance.
Regards,
Nic
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Re: [asterisk-users] How many lines do you use.
On 17:03, Wed 25 Nov 09, Robert Lister wrote: On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote: I use two ???lines??? though ???Line appearances??? would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) It does with the skinny firmware :) If you enable two line keys with the same user/pass then the phone will automatically put a second call/call waiting onto the second line key (assuming you have call waiting enabled.) But personally I preferred the way it presented the second call before, on a single line, and found the way it displays it with two lines a bit confusing. (I can't remember exactly why now, something like it would flash the second line icon but not show you the call information until that key was pressed, or you scrolled to it.) I could see users not getting on with this, so I didn't configure it like that. The rest can be used for speed dials, but these were of limited use to me since for some reason, although the line keys can be provisioned remotely over TFTP, the speed dials cannot. It's okay for personal use though. With the skinny firmware you configure the lines and speeddials in asterisk skinny.conf :) Personally moved off my 7960 in favour of the SNOM 370 as this supports far more features than the Cisco SIP image, which is only really a piece of migration fluff to enable Cisco to migrate customers away from competitors SIP systems onto Call Manager with the dual-boot/application loader. Asterisk has chan_skinny. The SNOM perhaps doesn't look as fancy as the Cisco handset, but it wins hands-down on SIP features. (the remote provisioning system was a little complicated to set up, but once set up it's okay.) It's a shame since the Cisco is a very capable (and expensive) handset, just let down by no development in the software other than small bug fixes for many years. If you dont like it, send the cisco to wedhorn or me so we can make chan_skinny even better. ;) -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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
[asterisk-users] Questions about static
Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc….. Typically a reboot of the phone resolves the problem…person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Strange…… Thanks --Dovey ___ -- 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] How many lines do you use.
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) It does with the skinny firmware :) The skinny channel driver also comes with the 'random crash' feature ;-p. But truth be told I only every tried chan_sccp2 (or was it b...). -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] How many lines do you use.
Hello all, Do you know if it IS possible to use multiple lines/extensions on SIP with a Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but have it register to a couple of different extensions, then use different ringtones to identify which line was ringing when a call came in. Thanks, Travis - Original Message - From: Michiel van Baak mich...@vanbaak.info To: asterisk-users@lists.digium.com Sent: Wednesday, November 25, 2009 2:40:07 AM Subject: Re: [asterisk-users] How many lines do you use. On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] How many lines do you use.
Travis Elsberry wrote: Hello all, Do you know if it IS possible to use multiple lines/extensions on SIP with a Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but have it register to a couple of different extensions, then use different ringtones to identify which line was ringing when a call came in. The grandstream gxp2000's have 4 lines, and this is what I did - I have 2 sip channels registered with the pbx, one for personal calls and one for business calls, so I can tell by which rings what type call it is, however what I did was use the 1st and 3rd channels since if I am on the line and another call comes in it rolls into the next line on the phone that way the roll happens to lines 2 and 4 that are not bound to an extension. if the roll happened to a channel that was registering rather than an empty button it would be confusing on the type of call. what is even more confusing is the 4 lines also roll into the speed dial / status buttons so if you have people on hold, it may look like you are getting a call from a speed dial link when its really an outside call. 4-6 lines sounds like a lot but I think in practice its more used for rollover than that many unique extensions on a phone. Thanks, Travis - Original Message - From: Michiel van Baak mich...@vanbaak.info To: asterisk-users@lists.digium.com Sent: Wednesday, November 25, 2009 2:40:07 AM Subject: Re: [asterisk-users] How many lines do you use. On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Please some enlightment on ENUM !!
Norbert Zawodsky skrev: SIP schrieb: Yes... you would have to register (and possibly pay for, dependent on the ENUM registrar) each individual number. The idea behind ENUM is that it's an E164 number that is already yours that maps to whatever you want it to map to (email, SIP, etc). The key point here is that you already own the E164 number. If you do, then you could register them all at e164.org for free. If you don't own the individual numbers, you shouldn't be allowed to register them as your own. That sort of breaks the ENUM concept of a number you take with you as a personal identifier. N. ___ -- 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 Hi N. ! Thanks for your answer. Either I don't understand what you want to tell me or this thread slowly drifts away from my original question. My original question was: If you own a telephone number which connects to your company and you have a PBX (like asterisk) and some extesniosn behind that, how/where do you enum-register each extension so that each extension can be reached from outside by a SIP uri? Meanwhile I managed to speak to a technician at my-enum.at, which is my registrar at e164.arpa. He *comfirmed* my original assumption: If you have a telephone number and want to paticipate in enum, you have to register that number at - for example - e164.arpa. If you operate extensions behind that number and you want them to be reachable too, you have to run your own DNS server and register this server at e164.arpa. This server is naturally under your responsibility and you manage all your extension yourself. It is works exactly like any other DNS resolution. But then you create phonenumbers in enum, which doesn't exist as pstn-numbers. Not the idea behind enum. On the other hand, if you owned 10 or 100 pstn-numbers in series, you could get the last one or two digits delegated to your dns-server. Leif ___ -- 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] Restricting transfers between SIP phones
Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or an original call. I've looked at the TRANSFER_CONTEXT setting, but that's only for transfers with # and the T and t flags to Dial(). But we use SIP hardphones (Linksys SPA942 Grandstream GXP2020), which have built-in transfer functions, and we would like to continue using those for transfers, rather than building it into features.conf or dialplan... Because we prefer attended transfers, and the user experience seems more modern. So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- 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] Questions about static
Is it a single user? Or every single phone? If it's a single user, and you can get hold of a UPS with power conditioning on it, try plugging the various devices into it - there might be some dirty power coming along. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Thursday, 26 November 2009 07:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about static Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc. Typically a reboot of the phone resolves the problem...person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Strange.. Thanks --Dovey IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Please some enlightment on ENUM !!
But then you create phonenumbers in enum, which doesn't exist as pstn-numbers. Not the idea behind enum. On the other hand, if you owned 10 or 100 pstn-numbers in series, you could get the last one or two digits delegated to your dns-server. Leif ___ -- 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 do I create numbers in enum which doesn't exist as pstn ? A simple example: My pstn number is +43-1-1234567. Everybody around the world can call me using this number. Lets say, I have 3 extensions: 0=reception, 10=secretary, 20=boss. If someone calls ENUMLOOKUP(+4311234567) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+43112345670) he will get a uri sip:0...@ip.of.my.asterisk ENUMLOOKUP(+431123456710) he will get a uri sip:s...@ip.of.my.asterisk or sip:1...@ip.of.my.asterisk (which ever you prefer) ENUMLOOKUP(+431123456720) he will get a uri sip:b...@ip.of.my.asterisk or sip:2...@ip.of.my.asterisk All this numbers exist because they connect to different persons. Why shouldn't that be the idea behind enum? Norbert ___ -- 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] 1.6.1.10 Music On Hold
Brilliant, thanks a lot. Best regards, Örn On Tue, Nov 24, 2009 at 1:39 PM, Santiago Gimeno santiago.gim...@gmail.comwrote: Hi, I think it can be related to https://issues.asterisk.org/view.php?id=16268 Best regards, Santi 2009/11/24 Örn Arnarson o...@arnarson.net Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- 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
[asterisk-users] Agent with External Number as Extension
Can you add an agent dynamically to a queue with an external number, i.e. cell phone as an extension? If so how? Thanks! ___ -- 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] Unable to open sound file error
Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when I try to dial the extension for test: [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File good does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open good (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for good -- Executing [...@default:12] BackGround(SIP/102-09b52260, evening ) in new stack [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File evening does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open evening (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for evening -- Executing [...@default:13] Hangup(SIP/102-09b52260, ) in new stack Any suggestions? Thanks in advanced for your help. ___ -- 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] 1950's UK rotary dial phone
Way back when, the wctdm driver needed a fix to make it more agreeable to pulse dials in the US. I suspect this is also the case in the UK. Speed and make break ratio are more critical , as pulse detection isn't nearly as smart as a PSTN exchange Search the wiki for more details. How that applies to the current set is unknown to me the suggested fix was never applied to the distributed code, and unfortunately there seems little interest in any support for pulse, or analog in general. John Novack Mike wrote: Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on the hook. However, using the rotary dialer does not work (works fine plugged into my phone line). I have read about the possibilty that the pulse settings may need adjusting in kernel.h in the dahdi driver but I have no idea what to set them to. I have tried tweeking them to various extents but I've not been able to bring it to life yet. Does anyone have any experience getting this to work? Does anyone know the specs for UK pulse dial? How long should the pulses be and what is the gap between them? Thanks, Mike. ___ -- 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 -- Dog is my co-pilot ___ -- 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] Questions about static
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote: Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc….. Typically a reboot of the phone resolves the problem…person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Only one user? Did you check to see if it is a bad handset cord? -chris www.mythtech.net ___ -- 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] Channel Variable
I believe ${IAXPEER(CURRENTCHANNEL)} should help you with the current
IAX2 name ... you can make DumpChan() to understand what kind of channel
variables you can use there.
--
razu
On 11/25/2009 09:57 PM, Nic Colledge wrote:
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel variable seems to be wrong.
I have been using Zoiper as my IAX client and Asterisk 1.6.2.0-rc6.
For the sake of debugging I have Verbose(1,Outgoing Call Handler
${CUT(CHANNEL,-,1)}) in the (internal - not default) dial plan.
Most of the time the channel variable is IAX2/10007 which is the desired
behaviour (with 10007 being the IAX username) but some of the time
IAX2/192.168.1.111:4569 is shown instead.
I would like to know why this is happening and if there is anything that can
be done to make it show the IAX2/10007 form every time?
I realise that I could use ${CDR(accountcode)} instead, and as it happens
this returns the correct account code value in both cases. However, I wanted
to be able to do this on a per-channel basis and multiple channels currently
share a common accountcode.
Any ideas what's going on here, is there something obvious I'm missing?
Thanks in advance.
Regards,
Nic
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