Re: [asterisk-users] AgentCallBackLogin vsAddQueueMember
I'd like to know what alternative is available for those who run a call centre with dynamic agent-queue allocation. We have people monitoring the queues and assigning agents depending on the queue demand. cheers! Santiago On 7/5/07, Martin Schrott - thinking:systems [EMAIL PROTECTED] wrote: sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be logged in. No need to create your own login functions. No additional tables, which members are logged in. Just one entry in queues.conf and agents.conf This is simple. For sure, it would also be possible to use addqueuemembers functionality: -make own tables where you save, in which queues each member has to be logged in. -create a table, to see wich members exist and which are logged in. Do not forget the destination to call them. -create a login functionallity, to use your tables. -Then add the member to each queue by calling aqm once for each queue. (Our cpu will thank us) for using it so much. -do not think of logs. (there are patches helping you... and members-name, wich you can use... try how) It is as simple as callbacklogin ;-) Next difficulty is, using agent-groups... When we use aqm to call different groups, we only have to make groups in agents.conf and put them into the queues. That is it. But no problem, we also can create additional tables and script a little bit. We do not need to sleep at night. To summerice: using aqm we would have to make own tables of groups. Then we have to make tables of members, that are logged in. Then we have to read this tables, check who is logged in, then call aqm for each member that is logged in and put it into each queue, the third table has saved this member for... !!! Only to write it here is more work then using agent callbacklogin! scripting it would be crazy, when callbacklogin does it for us !!! So we can only hope, that there will be an alternative application, that works like callbacklogin. I am sure, a lot of cc designers will stop upgrading, if callbacklogin is removed and now new simmilar application is provided! Nobody can effort to do this additional work to change all dialplans. :-) Where is the problem keeping callbacklogin as additional feature in future versions. Nobody has to support or change it. Just keep it working. Or create a new application that does all the same, when you can't stand it. If you can tell me in thre lines how to use addqueuemember doing all things we need from callbacklogin app, then I will use it from today on. Othervise it is a reinventing of the wheel. Hope there will be a alternate application in newer versions of asterisk. Thanks Martin - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Alan Ferrency [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 11, 2007 11:45 PM Subject: Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember Alan Ferrency wrote: However, this is not what we need. This adds a phone channel to the queue, and does not track which person is using that phone. This means that all queue activity is associated with a SIP channel in the logs, which is not acceptable. Right. This is why we added the 'membername' argument to the AddQueueMember application, so that queue logs can reflect a logical name for the queue member, regardless of what channel/interface they logged in from. Using this map of people to phones, our dial plan would then need to ensure that: - a person cannot be logged into more than one phone - only one person at a time can be logged into a phone - queue activity logs are associated with a person, not a phone For points #1 and #2, you are correct that this logic will have to be built. Point #3 is already taken care of by the addition of the 'membername' as I commented on above. However, I personally see this as a huge benefit; I much prefer Asterisk to provide mechanisms for users to do things, but not the policy on how they are to be used. When chan_agent is in use, you don't get to decide what to do if a second user tries to log in from the same channel, that has been decided for you. If instead you write that logic in the dialplan (or start from an example you find in the docs, on the wiki, etc.) you can completely control how the system behaves. Can the AddQueueMember solution handle the equivalent of autologoff if a queue member fails to answer a queued call in time? Absolutely; the example in doc/queues-with-callback-members.txt shows how to do it. To me, saying We
Re: [asterisk-users] Setting rxgain per channel
I'm sorry, I wanted to say FXO :P Thank you! On 3/30/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Delca [EMAIL PROTECTED] Date: Thu, 29 Mar 2007 18:39:37 -0300 How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS. Does FXS even use rxgain? To set rxgain for an FXO channel, simply put the entry before saying channel =. Hope this helps. Yuan Liu Thank you! Santiago del Castillo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting rxgain per channel
How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS. Thank you! Santiago del Castillo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Billing Plataforms
Hi, before you start throwing shoes to me, i know there are a lot of Asterisk Billing plataforms, but actually no one seems to accomplish what i need. They are to complex (a2billing) or doesn't have too much documentation (astbill and mcc) or are poorly developed (trabas). What i was looking for is a simple Asterisk billing plataform (with web based admin and customer interface) that only calculates the time of a call and the cost. For example let's suppose that a local extension calls to a UK number. What i need to know is the duration, and the total money spent on that call based on a dynamic tariff DB. I'd like the plataform to use an AGI script because it's a kind of postpaid/prepaid system. There shouldn't be any kind of authentification when the number is dialed. The Account number will be passed to the AGI as a parameter ( AGI(agiscript.agi|account) ) Somebody uses or used or is aware of something like this? Regards, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Interception
Hi, I'm deploying an asterisk PBX for a Call Center and i was ordered to check if the Customer Support Supervisor could intercept the calls so they can check how they employees work with Asterisk. I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i got a clue about intercepting calls. But actually i wanted to know if someone have experience with this sort of things. Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 and polycoms problem
didn't work :( Regards, Santiago On 9/20/06, Alyed Tzompa [EMAIL PROTECTED] wrote: Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it? hey... you have in your sip.conf configuration canreinvite=no... think this may be a problem: since Asterisk will always stay in the path of the RTPs, I think it might need to have the proper transcoder, as it does not, then the error arises... at least that's what I think :) set canreinvite=yes (or just comment it since that's the default) on both parties and try again. Let me know if it works. Alyed Return-Path: [EMAIL PROTECTED] Wed Sep 20 12:38:41 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 20 Sep 2006 12:38:41 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) Still having the same problem. i modified the sip.cfg in order to make g729 the first choice: voice.codecPref.G711A=3 voice.codecPref.G729AB=1 voice.codecPref.IP_4000.G711Mu=1 voice.codecPref.IP_4000.G711A=2 voice.codecPref.IP_4000.G729AB=/ Cheers, Santiago On 9/19/06, Alyed Tzompa wrote: Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec. Alyed Return-Path: Tue Sep 19 14:47:54 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Tue, 19 Sep 2006 14:47:54 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user 202 [EMAIL PROTECTED] secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dual asterisk, CallerID(name|number) problem
Hi, i'm having a bit of a problem with callerid(name|number). When a
call arrives to a server and it's forwarded to the other server, the
callerid(name|number) disapears.
Here's the output:
-- Executing NoOp(SIP/201.216.198.199-08171990, ) in new stack
-- Executing NoOp(SIP/201.216.198.199-08171990, 1151540837) in new stack
-- Executing Dial(SIP/201.216.198.202-081f19e0,
IAX2/asta/202|20) in new stack
this is when the call comes into the first asterisk. Here's what
happens with the second asterisk:
-- Executing NoOp(IAX2/astb-9, Unknown) in new stack
-- Executing NoOp(IAX2/astb-9, ) in new stack
-- Executing Macro(IAX2/astb-9, stdexten|202) in new stack
-- Executing Dial(IAX2/astb-9, SIP/202|20) in new stack
And the dial plans:
first:
exten = 01152584202,1,NoOp(${CALLERID(name)})
exten = 01152584202,n,NoOp(${CALLERID(number)})
exten = 01152584202,n,Dial(IAX2/asta/202,20)
exten = 01152584202,n,Hangup()
second:
exten = 202,1,NoOp(${CALLERID(name)})
exten = 202,n,NoOp(${CALLERID(number)})
exten = 202,n,Macro(stdexten,202)
exten = 202,n,Hangup
Cheers!
Santiago
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Re: [asterisk-users] g729 and polycoms problem
Still having the same problem. i modified the sip.cfg in order to make g729 the first choice: codecs preferences voice.codecPref.G711Mu=2 voice.codecPref.G711A=3 voice.codecPref.G729AB=1 voice.codecPref.IP_4000.G711Mu=1 voice.codecPref.IP_4000.G711A=2 voice.codecPref.IP_4000.G729AB=/ Cheers, Santiago On 9/19/06, Alyed Tzompa [EMAIL PROTECTED] wrote: Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec. Alyed Return-Path: [EMAIL PROTECTED] Tue Sep 19 14:47:54 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Tue, 19 Sep 2006 14:47:54 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user 202 [EMAIL PROTECTED] secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 and polycoms problem
Hi, I enabled sip debug and i get the following when i am trying to call a polycom phone with the same sip.cfg I sent before (with g729 as the primary codec): --- (15 headers 9 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.14.34.130 : 5060 (NAT) Found user '202' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.14.34.130:10008 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 20 16:52:57 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! Transmitting (NAT) to 10.14.34.130:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.14.34.130;branch=z9hG4bKa595a1daA5CED30F;received=10.14.34.130 From: Santiago del Castillo sip:[EMAIL PROTECTED];tag=FF5B5B3D-F2E725F8 To: sip:[EMAIL PROTECTED];user=phone;tag=as5e963058 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Capabilities: us - 0x100 (g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x0 (nothing) This line looks a little weird. As i understand peer should be the other phone and the other phone has g729 enabled at sip.conf (asterisk side) and sip.cfg (polycom phone side) And the line after that is what i get without sip debug Cheers! Santiago On 9/20/06, Delca [EMAIL PROTECTED] wrote: Still having the same problem. i modified the sip.cfg in order to make g729 the first choice: codecs preferences voice.codecPref.G711Mu=2 voice.codecPref.G711A=3 voice.codecPref.G729AB=1 voice.codecPref.IP_4000.G711Mu=1 voice.codecPref.IP_4000.G711A=2 voice.codecPref.IP_4000.G729AB=/ Cheers, Santiago On 9/19/06, Alyed Tzompa [EMAIL PROTECTED] wrote: Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec. Alyed Return-Path: [EMAIL PROTECTED] Tue Sep 19 14:47:54 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Tue, 19 Sep 2006 14:47:54 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user 202 [EMAIL PROTECTED] secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g729 and polycoms problem
Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user 202 [EMAIL PROTECTED] secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing Agent Function
Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missing Agent Function
Hi, Seems that FOP is a great tool and the person who made it is from my country :). But I'm having some problems configuring it. I made it possible to connect to the Asterisk as manager. Also I see a lot of output/input when I set debug=1. But, at the flash interface, the button that is under the arrow it's blinking... and as I can see in the official page demo, it isn't normal and I don't really know what could it be causing it. Cheers, Santiago On 8/31/06, Joe Dennick [EMAIL PROTECTED] wrote: The Flash Operator Panel (http://www.asternic.org/) can be configured to change the color of a phone's icon to indicate whether that agent is logged in or not. I've found it to be very useful and the agents don't mind using that to check their status as well as the queue status (how many callers are in the queue, etc.). Delca wrote: Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function since i need something to offer the agents a way to check if they are logged in or not. i was specting to use AGENT function for this. and i found out this: asterisk*CLI show function AGENT No function by that name registered. As i read here http://www.voip-info.org/wiki/view/Asterisk+functions . AGENT should be available for 1.2.x.x and i don't have it :( (chan_agent.so is loaded). Do i have to enable something else in order to use this function? or anyone else knows any other way to offer a way to check if an agent is logged in or not? (without using show agents, since it must be used phone-side and by agents). Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, two eth and two providers
Hi, I'm thinking on setting up an asterisk server with two providers. One will let us to make international calls and provide to us a TollFree number. The other will provide local numbers (i'm from Argentina). The problem is that the local number provider requires a dedicated connection and the asterisk server is behind a NAT. There's no problem with the first provider, i just forward the ports and modify the required firewall rules and it's done. The problem comes with the second provider. They gave to me an IP, GW, nmask, etc because it's an static IP. No problem with that.. i configured eth1 with that information.. and works great. The problem comes when setting up asterisk because i just can set one externip in sip.conf file. Also, the packets are beign forwarded to the server (i checked this with TCPDump) but asterisk doesn't give an answer at all. INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Does asterisk support this kind of setup? i mean two providers (one behind NAT and the other with a dedicated connection) with thifferent eth controllers. If you have questions or doubts about what i said. Feel free to ask i'm really looking forward to solve this problem. Best regards, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch
Bu! :( But if i have the 'special cable' and i connect one end point to the phone and the other to a female rj45 and from the female to the PoE Switch there's a normal cable.. Is it going to work? Thanks! Santiago On 8/8/06, BJ Weschke [EMAIL PROTECTED] wrote: On 8/7/06, Santiago del Castillo [EMAIL PROTECTED] wrote: Hi, someone has tried this combo? I have a SRW224P switch and i tried to make the phone to work with PoE on this switch but it isn't work. I read about this and i found that this phone needs an 'special cable' in order to work with PoE. It's that true? Isn't there any way to make it work with a normal cable? :( No. You must have the cable for the 301's and 501's. These phones are not native PoE devices. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Dial Tone
Hi, i'm having problems with DTMF, the problems are with established connections and some IVRS. When i call to other number which has an IVR, some digits doesn't work. I digit a long number (required by the IVR, at least a 10 digit number) and it doesn't work. I think it's about DTMF signalling, i've all my extensions with RFC2833 mode, i've an LinkSys PAP-2 and a Polycom 301, allowed codec is ulaw. If you need more information, pleas feel free to ask :) Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Overriding # at the end
Hi Jay, thanks for the help, it was really useful :) I realized that my extensions.conf was a mess and i re-do it and modified the ATA dial plan and now it's more structured and scalable. Thanks! Santiago On 7/21/06, Jay Milk [EMAIL PROTECTED] wrote: Delca wrote: Fixed, i'm the kind of guy who ask and later find the solution :$ it is a Linksys PAP-2 ATA setting in Regional - Control Timer Values - Interdigit Long Timer (this is in advanced mode). Sorry :) Santiago On 7/20/06, Delca [EMAIL PROTECTED] wrote: Hi, I'm using a Linksys PAP-2 to test my next PBX with asterisk. The problem I'm haivng is that when I dial the extension, I've to end it with # and then it starts calling is there any way to override that # so with just dialing the 3-digit extension I'll be able to call? This actually works great with a voipjet configuration I already have .. when I dial an US number (i.e.: 12245684486 ) it starts dialing that number. But if I do the same with an extension, I just have to wait until i press #. I just want to dial 123 :( Cheers! Santiago That's probably not the best solution. You may want to look at dial-plans here. For example, I have this one: (*xxS0|011x.|1xxS0|2xxS0|6xxxS0|7xxS0|8.|911S0) *xx are services and are immediately called 011x. allows for international numbers 1xxS0 makes sure 1+10 digit US numbers are called instantly 2xxSO makes sure extensions (three digits, all beginning with 2) are dialed instantly etc... don't forget 911S0 -- this dials 911 immediately as well. Check the sipura website for config info on dial plans. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get blind transfer to work
Oops! :P nope, i didn't! It was the problem :$ Thank you!! Santiago On 7/20/06, C F [EMAIL PROTECTED] wrote: You using the t or T options in the dial app? On 7/19/06, Delca [EMAIL PROTECTED] wrote: Hi, Now that i fixed the problem with roundrobin, now i can't get Blind Transfer to work. I already tried to modify blindxfer option in features.conf with almost any number and still doesn't work. When i dial an extension. I pick up the phone, and then i press # to transfer the call and nothing happens, i can hear the # tone in the other phone. Somebody had the same problem? I need to do a blind transfer in order to do a conference. Anyone has any other options or conference config? i'm trying to follow this instrucions: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macroview_comment_id=11271 but i can't continue if Blind Transfer doesn't work :( Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Overriding # at the end
Hi, I'm using a Linksys PAP-2 to test my next PBX with asterisk. The problem I'm haivng is that when I dial the extension, I've to end it with # and then it starts calling is there any way to override that # so with just dialing the 3-digit extension I'll be able to call? This actually works great with a voipjet configuration I already have .. when I dial an US number (i.e.: 12245684486 ) it starts dialing that number. But if I do the same with an extension, I just have to wait until i press #. I just want to dial 123 :( Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Overriding # at the end
Fixed, i'm the kind of guy who ask and later find the solution :$ it is a Linksys PAP-2 ATA setting in Regional - Control Timer Values - Interdigit Long Timer (this is in advanced mode). Sorry :) Santiago On 7/20/06, Delca [EMAIL PROTECTED] wrote: Hi, I'm using a Linksys PAP-2 to test my next PBX with asterisk. The problem I'm haivng is that when I dial the extension, I've to end it with # and then it starts calling is there any way to override that # so with just dialing the 3-digit extension I'll be able to call? This actually works great with a voipjet configuration I already have .. when I dial an US number (i.e.: 12245684486 ) it starts dialing that number. But if I do the same with an extension, I just have to wait until i press #. I just want to dial 123 :( Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't get blind transfer to work
Hi, Now that i fixed the problem with roundrobin, now i can't get Blind Transfer to work. I already tried to modify blindxfer option in features.conf with almost any number and still doesn't work. When i dial an extension. I pick up the phone, and then i press # to transfer the call and nothing happens, i can hear the # tone in the other phone. Somebody had the same problem? I need to do a blind transfer in order to do a conference. Anyone has any other options or conference config? i'm trying to follow this instrucions: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macroview_comment_id=11271 but i can't continue if Blind Transfer doesn't work :( Cheers! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue RoundRobin
Hi Kevin, thanks for answering. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? Yes, both agents are in the same group. If you make the one agent busy does it ring to the next phone? Nope If not, what does the CLI say when it tries to connect the next call to the second phone? Here's the URL with complete IVR procedure with 2 agents online: http://pastebin.com/750304 Regards, Santiago On 7/17/06, Kevin Smith [EMAIL PROTECTED] wrote: Hi Santiago, Unless it is a typo on the wiki, I think you want your queue.conf to be like this: member = Agent/@1 member = Agent/:2,1 That way you include group 1, and then include group 2 with consideration of penalty. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? If you make the one agent busy does it ring to the next phone? If not, what does the CLI say when it tries to connect the next call to the second phone? Kevin Santiago del Castillo wrote: Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue RoundRobin
The only way i figured out to fix this problem was by setting autologoff lower than Dial timeout. This way if the agent doesn't answer, it will log off before de Dial timeout So the next phone to ring will be the next available agent. Cheers, Santiago On 7/17/06, Delca [EMAIL PROTECTED] wrote: Hi Kevin, thanks for answering. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? Yes, both agents are in the same group. If you make the one agent busy does it ring to the next phone? Nope If not, what does the CLI say when it tries to connect the next call to the second phone? Here's the URL with complete IVR procedure with 2 agents online: http://pastebin.com/750304 Regards, Santiago On 7/17/06, Kevin Smith [EMAIL PROTECTED] wrote: Hi Santiago, Unless it is a typo on the wiki, I think you want your queue.conf to be like this: member = Agent/@1 member = Agent/:2,1 That way you include group 1, and then include group 2 with consideration of penalty. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? If you make the one agent busy does it ring to the next phone? If not, what does the CLI say when it tries to connect the next call to the second phone? Kevin Santiago del Castillo wrote: Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue RoundRobin
I fixed the problem by listing all agents 1 by 1. I think this is one thing that should be fixed. Or at least Agent/@1 should work with roundrobin. Cheers, Santiago On 7/17/06, Delca [EMAIL PROTECTED] wrote: The only way i figured out to fix this problem was by setting autologoff lower than Dial timeout. This way if the agent doesn't answer, it will log off before de Dial timeout So the next phone to ring will be the next available agent. Cheers, Santiago On 7/17/06, Delca [EMAIL PROTECTED] wrote: Hi Kevin, thanks for answering. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? Yes, both agents are in the same group. If you make the one agent busy does it ring to the next phone? Nope If not, what does the CLI say when it tries to connect the next call to the second phone? Here's the URL with complete IVR procedure with 2 agents online: http://pastebin.com/750304 Regards, Santiago On 7/17/06, Kevin Smith [EMAIL PROTECTED] wrote: Hi Santiago, Unless it is a typo on the wiki, I think you want your queue.conf to be like this: member = Agent/@1 member = Agent/:2,1 That way you include group 1, and then include group 2 with consideration of penalty. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? If you make the one agent busy does it ring to the next phone? If not, what does the CLI say when it tries to connect the next call to the second phone? Kevin Santiago del Castillo wrote: Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
