[asterisk-users] dahdi-channels.conf vs. chan_dahdi.conf
Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with, apparently, different functionality, but very similar names?) * If I'm getting power to my analog phones, but no dial tone, which file should I be editing? * Likewise (and almost certainly related) if dahdi_cfg shows the channels, but dahdi show channels only shows my T1 spans, which file should I be editing? Could someone point me to some sample analog configs? Most of my searches have wound me up with GUI folks, and I'm just doing good ol-fashioned hand editing on an Ubuntu system. Thanks! -Ken -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- 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] dahdi-channels.conf vs. chan_dahdi.conf
hi, chan_dahdi is some kind of generic config file, and dahdi-channels the config file where you configure your channels... so to say hardware specific. dahdi-channels.conf is normally a generated file which in turn is included by chan-dahdi... it makes sense to me to divide dahdi channel config in two files... but as one of them is included by the other you could merge them by hand... but remember you have then to edit it yourself if your hardware configuration changes (e.g. after adding a new card, as it was in your case...) so if your analog card requires drivers, install them or look in you /etc/dahdi/modules if you disabled the loading of the module for your newly added card. after this run dahdi_genconf and all should be set up atomagically... regards, yves Am 28.03.2013 14:44, schrieb Ken D'Ambrosio: Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with, apparently, different functionality, but very similar names?) * If I'm getting power to my analog phones, but no dial tone, which file should I be editing? * Likewise (and almost certainly related) if dahdi_cfg shows the channels, but dahdi show channels only shows my T1 spans, which file should I be editing? Could someone point me to some sample analog configs? Most of my searches have wound me up with GUI folks, and I'm just doing good ol-fashioned hand editing on an Ubuntu system. Thanks! -Ken -- _ -- 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] dahdi-channels.conf vs. chan_dahdi.conf
- Original Message - From: Ken D'Ambrosio k...@jots.org Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with, apparently, different functionality, but very similar names?) dahdi-channels.conf is generated by dahdi_genconf so as to not accidentally overwrite chan_dahdi.conf. You should #include dahdi-channels.conf inside of dahdi_genconf. You can use #include's in all Asterisk config files. Various asterisk distributions break out conf files into multiple files using includes for compartmentalization and ease of user customization. * If I'm getting power to my analog phones, but no dial tone, which file should I be editing? You should be editing whichever file contains the options you need to edit. You shouldn't have duplicated settings between the two files - if you do, you should resolve that. * Likewise (and almost certainly related) if dahdi_cfg shows the channels, but dahdi show channels only shows my T1 spans, which file should I be editing? You should make sure the missing channels are configured appropriately in either chan_dahdi.conf or dahdi-channels.conf files. Could someone point me to some sample analog configs? Most of my searches have wound me up with GUI folks, and I'm just doing good ol-fashioned hand editing on an Ubuntu system. http://docs.digium.com/TDM410/analog410series_manual.pdf see pages 38-44 Hope those help! -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.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] Pattern matching repeating digits
You are correct, it is stupid 8-)
exten = 233,1,Goto(dial-out,${EXTEN},1)
exten = 255,1,Goto(dial-out,${EXTEN},1)
[dial-out]
exten = _XXX,1,DoStuff()
exten = _XXX,n,AndMoreStuff()
exten = _XXX,n,Dial(something)
exten = _XXX,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nathan Anderson
Sent: Wednesday, March 27, 2013 2:18 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Pattern matching repeating digits
'lo, all,
Is there some (possibly undocumented?) way that I can pattern-match on a
specified number of repeating digits? (Something similar to regular
expressions' {})
Here's an example: let's say I have a string of things that need to be done for
both extensions 233 and 255. I can either...
A) Repeat the exact same code for both extensions, like so:
exten = 233,1,DoStuff()
exten = 233,n,AndMoreStuff()
exten = 233,n,Dial(something)
exten = 255,1,DoStuff()
exten = 255,n,AndMoreStuff()
exten = 255,n,Dial(something)
...which is stupid, or...
B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as
specific of a match as possible, like so:
exten = _2[35][35],1,DoStuff()
exten = _2[35][35],n,AndMoreStuff()
exten = _2[35][35],n,Dial(something)
...but this will not only match 233 and 255, but 235 and 253 as well.
It'd be nice if there was a substitute character that meant a character that
is exactly the same as the preceding one; for example, if R was meant to
represent such a concept, then this would do what I want:
exten = _2[35]R,1,DoStuff()
exten = _2[35]R,n,AndMoreStuff()
exten = _2[35]R,n,Dial(something)
You could even do crazy things like chain them together (this would match 2
and 2 and nothing else);
exten = _2[35]RRR,1,DoStuff()
exten = _2[35]RRR,n,AndMoreStuff()
exten = _2[35]RRR,n,Dial(something)
Am I missing something or does this really not exist?
Thanks,
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
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Re: [asterisk-users] Dedicated LCR Solutions
I know Evariste Systems has a product called CSRP - http://evaristesys.com/pub/CSRP-ProductOverviewCapabilitiesSurvey.pdf - that looks very interesting and it is built for high-volume scenarios. It is basically a standalone box you route calls to. Just my two cents, l. 2013/3/26 Nick Khamis sym...@gmail.com Hello Everyone, Was wondering what some of you for stand alone LCR implementations. I am aware of the LCR module within asterisk and a2billing however, we are looking for a standalone self less coupled solution. Not sure if such thing exist. Kind of like CDR Tool but for LCR... Thanks in Advance, Nick -- _ -- 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
[asterisk-users] Asterisk 1.8.21.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.21.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix issue where chan_mobile fails to bind to first available port (Closes issue ASTERISK-16357. Reported by challado) * --- Fix station ringback; trunk hangup issues in SLA (Closes issue ASTERISK-20462. Reported by dkerr) * --- Fix Queue Log Reporting Every Call COMPLETECALLER With h Extension Present (Closes issue ASTERISK-20743. Reported by call) * --- Fix Record-Route parsing for large headers. (Closes issue ASTERISK-20837. Reported by Corey Farrell) * --- Fix AMI redirect action with two channels failing to redirect both channels. (Closes issue ASTERISK-18975. Reported by Ben Klang) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.21.0 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 11.3.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix issue where chan_mobile fails to bind to first available port (Closes issue ASTERISK-16357. Reported by challado) * --- Fix Queue Log Reporting Every Call COMPLETECALLER With h Extension Present (Closes issue ASTERISK-20743. Reported by call) * --- Retain XMPP filters across reconnections so external modules continue to function as expected. (Closes issue ASTERISK-20916. Reported by kuj) * --- Ensure that a declined media stream is terminated with a '\r\n' (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) * --- Fix pjproject compilation in certain circumstances (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0 Thank you for your continued support of Asterisk! -- _ -- 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] To queue or not to queue...
Hello All, History ~ I recently took a position with a call center. At the time they had about 50 agents in a call queue. The queue was setup to ringall. The agents use Eyebeam softphones. Everything is local lan, no routers, everything connected via Cisco 3600 10/100 switches. Now we are up to about 150 agents, and I have kept everything pretty much the same way for a couple of reasons. However, those reasons are slowly drifting away and it's become the right time for me to start questioning some of the previous configuration. Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Point of discussion~ Is there really a reason to maintain a queue? With the companies growth they are now discussing the option of sending certain affiliates to certain sales reps. Am I better off using ring groups? Additionally I am working towards running as much of my configs via mysql as possible and turning up multiple servers to handle the calls. So far we have reached 130 simultaneous calls on one server, and about 10,000 calls processed during a 12 hour day. Thanks for reading. I look forward to hearing peoples views on this... -- _ -- 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] To queue or not to queue...
On Thu, Mar 28, 2013 at 12:55 PM, Gregory Malsack gmals...@coastalacq.comwrote: Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Which version of Asterisk? Have you looked for solutions to the root problem? I don't run any servers with that many agents, but have never run into issues like this with a few dozen. Large ring groups can become unwieldy and problematic themselves. There's also a limit to how long the entire dial string can be, though I can't remember what that size is. You said everything is on a LAN, but have you looked at the possibility of issues between switches? Can you examine the logs of bad calls and see if the failures happen on a specific switch in the network, or other correlation like that? Do you use VLANs or layer 3 switching? -- 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
Re: [asterisk-users] To queue or not to queue...
Asterisk version 1.8.20.1 Already checked the switches, no noteworthy port issues. no vlans used or layer 3 switching. On 03/28/2013 03:18 PM, Carlos Alvarez wrote: On Thu, Mar 28, 2013 at 12:55 PM, Gregory Malsack gmals...@coastalacq.com mailto:gmals...@coastalacq.com wrote: Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Which version of Asterisk? Have you looked for solutions to the root problem? I don't run any servers with that many agents, but have never run into issues like this with a few dozen. Large ring groups can become unwieldy and problematic themselves. There's also a limit to how long the entire dial string can be, though I can't remember what that size is. You said everything is on a LAN, but have you looked at the possibility of issues between switches? Can you examine the logs of bad calls and see if the failures happen on a specific switch in the network, or other correlation like that? Do you use VLANs or layer 3 switching? -- 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] To queue or not to queue...
On Thu, 28 Mar 2013 14:55:45 -0500 Gregory Malsack gmals...@coastalacq.com wrote: History ~ I recently took a position with a call center. At the time they had about 50 agents in a call queue. The queue was setup to ringall. The agents use Eyebeam softphones. Everything is local lan, no routers, everything connected via Cisco 3600 10/100 switches. Now we are up to about 150 agents, and I have kept everything pretty much the same way for a couple of reasons. However, those reasons are slowly drifting away and it's become the right time for me to start questioning some of the previous configuration. Have you considered switching the strategy to roundrobin or leastrecent? You could give it a very low agent timeout (like 5 seconds), so the caller doesn't have to wait long if it has to ring a few people. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Asterisk 11 -CDR values changed in hangup handler not saved ?
Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? Regards -- _ -- 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] To queue or not to queue...
Hii, If you like to use ringall strategy only, then better use different ringgroup to fulfill your purpose. However, as a callcenter aspect, you should think of roundrobin or leastrecent strategy. It will solve your purpose also and give a better performance too by resources i.e. hardware and agents. Regards, Bharat Lalcheta On Fri, Mar 29, 2013 at 4:41 AM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 28 Mar 2013 14:55:45 -0500 Gregory Malsack gmals...@coastalacq.com wrote: History ~ I recently took a position with a call center. At the time they had about 50 agents in a call queue. The queue was setup to ringall. The agents use Eyebeam softphones. Everything is local lan, no routers, everything connected via Cisco 3600 10/100 switches. Now we are up to about 150 agents, and I have kept everything pretty much the same way for a couple of reasons. However, those reasons are slowly drifting away and it's become the right time for me to start questioning some of the previous configuration. Have you considered switching the strategy to roundrobin or leastrecent? You could give it a very low agent timeout (like 5 seconds), so the caller doesn't have to wait long if it has to ring a few people. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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 -- Bharat Lalcheta -- _ -- 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
