Re: [asterisk-users] I need a second opinion on a new phone system deployment
Please also have a look at the gateway boxes from berofix (http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but have used different products from them over last few yeas and all have survived and are stable. Documentation is open and free on their wiki. They provide updates. They are not the cheapest but they have different vendors and they are sold in online webshops. You can choose for the inside PCI(e) cards or their external boxes. Last few years I went for the external boxes. They can be fitted in a server rack or you mount them against the wall with screws. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block certain numbers
On Monday 17 June 2013, vortex wrote: That seems suitable. for the cid name i am using asterisk's internal database for the lookup such as *database put cidname 222333 xyzwhoevername* then how do you create the database for the check_ban_db? However you like! I'd use a MySQL database, and my check_ban_db and add_to_ban_db scripts would be written in Perl or Bash. But that's just because that's what I'm familiar with. If you prefer Postgres and Python or PHP, or even flat text files and awk, by all means use that. Paid-for stuff isn't welcome here, so you will have to contact me off-list if you want me to write it for you . -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Limit Callers
On 17 June 2013 11:02, Shanavaz E A shanava...@yahoo.com wrote: Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I am not aware of an existing way to do this. By default, Asterisk does not appear to be able to dynamically change the priority of a queue, which seems to be what you are after. On one of my systems I implemented the patch at https://issues.asterisk.org/jira/browse/ASTERISK-17570 which, although it does not do exactly what you are after, it does help a great deal. Basically it dynamically increases the priority of callers based on their hold time across all queues, so that for a given agent they will be presented with the call having the longest wait time across all the queues they are a member of. This has made a big difference to our avg hold time, as queues are no longer competing against one anther for available agents. Hope this helps. -Barry Flanagan I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. Regards Shanavaz. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Limit Callers
You should have different sets of agents logged in to different queues and you should have a monitor to move them from one queue to the other based on incoming traffic. l. 2013/6/17 Shanavaz E A shanava...@yahoo.com Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. Regards Shanavaz. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Limit Callers
Hello Shanavaz., Please find some quick thoughts: * 2 main queues * agents logged on one or on both main queues * before sending a new call to one of the main queues check the number of waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for 30 sec) the call on a empty members queue/parking slot/music-on-hold if the queue threshold is reached. The threshold could be read from a database, internal astdb or could be set as a global variable updated when agents login/logout/pause/unpause or could be dynamically computed based on QUEUE_MEMBER_COUNT / QUEUE_MEMBER_LIST After the divert period is ended the call will return and the threshold is checked again, etc. This method have some negative impacts (the entry position number for calls over the threshold //origposition// will have no meaning, a newer call could be served before an older one, etc.) but you could manipulate the call flow exactly how you want. HTH, Ioan http://www.modulo.ro On Tue, Jun 18, 2013 at 12:05 PM, Lenz Emilitri lenz.lo...@gmail.comwrote: You should have different sets of agents logged in to different queues and you should have a monitor to move them from one queue to the other based on incoming traffic. l. 2013/6/17 Shanavaz E A shanava...@yahoo.com Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. Regards Shanavaz. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
My experience was good, Nicolas was very helpful and quick Regards, Zohair Raza On Tue, Jun 18, 2013 at 4:26 AM, Carlos Alvarez car...@televolve.comwrote: No vacation notice, nothing, other than the system auto-replying saying that the ticket will be closed because we didn't have any action on it. Rather distressing for our customers. On Mon, Jun 17, 2013 at 5:22 PM, Gregory Malsack gmals...@coastalacq.comwrote: No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for fop2 a month or so ago. Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto dialer scripts and software
I think political calling is less restrictive than you think! This is three years old, but probably mostly applicable. http://www.ilga.gov/commission/lru/Feb2010FirstRdg.pdf --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cjwstudios Sent: Thursday, May 23, 2013 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Auto dialer scripts and software As long as you're dialing a screened registered voter list and don't call .gov or .edu, you're fine. On Wed, May 22, 2013 at 5:54 AM, Don Kelly d...@donkelly.biz wrote: Calls on behalf of political candidates are generally legal--even to people on the do not call lists. It doesn't seem to be possible to pass legislation preventing them. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Wednesday, May 22, 2013 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Auto dialer scripts and software On 22/5/13 10:54 am, A J Stiles wrote: You do know that sort of thing is against the law -- or at least requires a permit from the authorities -- in most civilised countries, right? And it's worth adding that even if it is legal in your country, you're almost guaranteed to offend/annoy your target audience. Recorded calls always do. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended transfer problem
I have a setup where there are occasional problems with attended transfers. I have already checked the devices as well as the relevant DTMF modes (SIP INFO and rfc2833). I could not find any problems here. The setup is a follows: The front desk (F) accepts calls from customers (C). In some cases F needs to transfer C to a specific department (D). If D cannot handle the problem, D tries to transfer to a specialist S. The problem sometimes occurs when D tries an attended transfer for C to S. The description I have so far is that Asterisk sometimes does not seem to accept DTMF signals. Is it conceivable that that this is a re-INVITE/directmedia problem? Since I have only limited access to the system, I'd like to rule out some causes. I am also not sure if the transfer from F to D is handled via Asterisk using DTMF signalling, or whether F uses the SIP phone's capabilities for the transfer, so that Asterisk might already not know anything about D. Except for C, everybody is inside the same subnet. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Identify port on Khomp card.
Greetings. I've plugged 3 analog lines on an ethernet cable in an Khomp card to receive it's incoming calls. Without any configuration, when I call those numbers the asterisk server automatically answer the call and play the default music. The problem is: I need to discern the lines and redirect each one to his respective extension. Since they doesn't got any Caller ID Service the asterisk cannot distinguish them and give the default treatment. How do I identify the originating port of the call, the originating line or somehow discern each line to make custom configuration for them? My environment: - Elastix v2.3 - Khomp KFXO IP - All the lines are attached to the card on the ethernet interface. -- Att.* *** Luis H. Forchesatto -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] language specific email templates
Hi, I am new to Asterisk. I'm using it behind a kamailio sip-router to provide voicemail boxes to sip-users. I followed these instruction: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x to set everything up, using ARA with a MySQL DB. After a few tweaks everything is basically working, however, a few questions remain that I could not find clear answers for - anywhere. Maybe some of the experts can help a little ... So far, my Asterisk DB features 1 table (voicemessages) and 2 views (vmusers and sipusers). vmusers shows the following columns: uniqueid, customer_id, context, mailbox, password, fullname, email, pager, stamp (sipusers shows: name, defaultuser, type, secret, host, callerid, context, mailbox, nat, qualify, fromuser, fromuser, authuser, fromdomain, insecure, canreinvite, disallow, allow, restrictcid, defaultip, ipaddr, port, regseconds) Questions: 1) In vmusers, the content for column fullname is composed from 2 columns (i.e. first_name, last_name) of the kamailio subscriber table that do not even natively exist there, but were manually added by me (to comply with the instructions). Q1a: Is the fullname column in the vmusers view mandatory for Asterisk to function properly? fullname might always stay empty... Q1b: Is the stamp column in the vmusers view mandatory for Asterisk to function properly? 2) in my world, sip-users have a language. At certain points, Asterisk will communicate with my users regarding voicemail. e.g.: a) Asterisk delivers the message left as a wav-attachment by email, b) VoicePrompts guide the user through the voicemail-menu So far, I am able to hard-code the language for the voiceprompts in extension.conf with Set(CHANNEL(language)=fr). But this set the language for everybody. Also, you can customise the email template ($emailbody, etc) for voicemail delivery in voicemail.conf. However, this is also covers one language only. Q2a:How can configure Asterisk I pick the voicemail-prompts of the respective sip-users language? Q2b:Is it useful for that purpose, to add a language column to the vmusers view ? Q2c:Where can language-specific email-templates for voicemail delivery be supplied and how can be achieved that the correct one is used ? 3) I would also like to use Asterisk as a SIP client (since Kamailio can't do this). Each vmuser may have one or many SIP accounts with foreign registrars. I would like Asterisk to register those accounts with their registrars and forward incoming calls to the kamailio (parent) account that the vmuser owns. (The kamailio (parent) account -in turn- shall forward this call to the user's asterisk voice-box, if the user is busy/not-registered/etc ). Q3a:How can I use ARA to configure Asterisk as a SIP client to act in the described fashion ? Q3b:Is it useful/necessary for that purpose, to host dial-plans in the MySQL-DB as well ? Q4: Can you point to some documentation that explains ARA a little more in depth and possibly illustrates a few examples? For any hint - thank you very much in advance! Best regards, -Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
