Re: [asterisk-users] Know who's logged in
Hello Mark , On Fri, 27 Mar 2009, Mark Michelson wrote: Mr. James W. Laferriere wrote: On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL Just typing the word agent will result in the message you see. If you press the tab key after typing the word agent you should see that one of your options for completing the command is agent show. This command is definitely in all releases of 1.4. I specifically double-checked and the command works fine for me in 1.4.22. Mark Michelson asterisk-2*CLI help agent No such command 'agent'. asterisk-2*CLI agent No such command 'agent ' (type 'help agent' for other possible commands) Maybe I've mis-configure my compile options or something but ... Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] hum noise
HI, We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango ___ -- 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 to Integrate Neospeech with Asterisk
Please let me know in brief, How the TTS interface work with asterisk? I call that from Asterisk using AGI So does calling it from AGI, return the converted autdio file? OR the audio files are generated well before, and calling from AGI will use that pre-generated files? It will be great that calling NewSpeech from AGI, should return with created wave file. Do you think is it possible with NeoSpeech. If yes let me know brief working idea behind this. Thanks msp ___ -- 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 to Integrate Neospeech with Asterisk
Please let me know in brief, How the TTS interface work with asterisk? I call that from Asterisk using AGI So does calling it from AGI, return the converted autdio file? OR the audio files are generated well before, and calling from AGI will use that pre-generated files? It will be great that calling NewSpeech from AGI, should return with created wave file. Do you think is it possible with NeoSpeech. If yes let me know brief working idea behind this. ___ -- 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] ITSP's no longer supporting IAX?
Steve Totaro wrote: I understand you are a developer and you want IAX2 to be great. That is your job, but the fact is that it is not and has caused audio and security problems for YEARS in EVERY release. It should bug you and everyone at Digium that waves the IAX2 flag. Can you elaborate on these audio and security problems Steve? Looking at the two protocol specs I cannot see a basis for your claim. IAX doesn't embed the local IP address in the packet data but that's surely no substantive security. It does separate data and signaling at the application-level, but again, that's no basis for such a claim. Protocols must be looked at separately from their implementations. From the various responses it appears that Asterisk 1.4's implementation of IAX has flaws. These do not necessarily reflect on the protocol. OTOH, there are a lot of engineers with SIP skill and experience who, naturally, are concerned with their investment in time, education, and experience. While this may or may not apply to Sonicwall engineering, it's also true that any streaming protocol will be better handled by devices that process packets in ASICs (high-end firewalls) rather than CPUs (PCs and low-end firewalls). FWIW (2 data points) I get uniformly better service from our IAX trunk provider than our SIP trunk provider. No idea whether that's protocol, implementation (1.4 on my side), or provider-related though I suspect the later. Roger Marquis ___ -- 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] ITSP's no longer supporting IAX?
Roger Marquis wrote: Steve Totaro wrote: I understand you are a developer and you want IAX2 to be great. That is your job, but the fact is that it is not and has caused audio and security problems for YEARS in EVERY release. It should bug you and everyone at Digium that waves the IAX2 flag. Can you elaborate on these audio and security problems Steve? Looking at the two protocol specs I cannot see a basis for your claim. IAX doesn't embed the local IP address in the packet data but that's surely no substantive security. It does separate data and signaling at the application-level, but again, that's no basis for such a claim. Protocols must be looked at separately from their implementations. From the various responses it appears that Asterisk 1.4's implementation of IAX has flaws. These do not necessarily reflect on the protocol. OTOH, there are a lot of engineers with SIP skill and experience who, naturally, are concerned with their investment in time, education, and experience. While this may or may not apply to Sonicwall engineering, it's also true that any streaming protocol will be better handled by devices that process packets in ASICs (high-end firewalls) rather than CPUs (PCs and low-end firewalls). This sounds like a bunch of gobbledegook spewed out by those very high end firewall vendors. Call it what you want but anything that processes packets in any way and makes a decision on what to do is by definition a CPU. And a general purpose CPU is not exactly poor at the job. If you look at utilization levels and latency on a typical CPU you would have thrown away already as a server, its barely even noticable utilization running a complex set of rules on a high volume data stream. FWIW (2 data points) I get uniformly better service from our IAX trunk provider than our SIP trunk provider. No idea whether that's protocol, implementation (1.4 on my side), or provider-related though I suspect the later. Roger Marquis ___ -- 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] Need to find small footprint asterisk platform
There's a couple of manufacturers offering smaller versions (than four ports) of the OpenPBX design: EdgePBX FX02: http://www.edgepbx.cn/shop/index.php?controller=productpath=19product_id=1 Atcom IP01: http://www.atcom.cn/En_products_IP01.htm Atcom IP02: http://www.atcom.cn/En_products_IP02.htm None have USB or wifi, as far as I know. If you can elaborate on what the USB is needed for, maybe someone can suggest other ways to achieve the same end. -- Paul Robin Rodriguez wrote: what about http://www.rowetel.com/ucasterisk/ip04.html seems like what you might be after good luck snip ___ -- 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] hum noise
On Sat, 28 Mar 2009, Rilawich Ango wrote: We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango It depends on the source :) What computer, what type of connection (Zap (I'm a 1.2 Druid), SIP, IAX), what interface hardware (none, t100p, etc.)? Anything else like the channel bank sits atop the isolation transformer for the entire building? I built an Asterisk server out of an old Fiire Station. It's Via micro-atx in a shoe-box and a tdm400. Having everything crammed into such a small space put the horizontally mounted tdm card millimeters above the CPU. The noise during dialtone was so bad I scrapped the project there and then. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
Jon Pounder wrote: This sounds like a bunch of gobbledegook spewed out by those very high end firewall vendors. Call it what you want but anything that processes packets in any way and makes a decision on what to do is by definition a CPU. You won't find much support for that opinion in network engineering circles. The processing advantage of ASICs is easily measured and widely documented. ASICs are particularly critical to latency-sensitive protocols and those using small packet sizes with correspondingly high packet counts. According to Praveen Kumar (Founder/CEO of Packet Island) the ASIC differential is even more noticeable with interactive streaming video than streaming audio. Roger Marquis ___ -- 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] ITSP's no longer supporting IAX?
Roger Marquis marq...@roble.com writes: ASICs are particularly critical to latency-sensitive protocols and those using small packet sizes with correspondingly high packet counts. According to Praveen Kumar (Founder/CEO of Packet Island) the ASIC differential is even more noticeable with interactive streaming video than streaming audio. However, there is nothing that makes IAX more difficult than RTP. For both, you just have to use the standard UDP forwarding path without doing inspection. ASIC's won't care either way. /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] oh323 to h323
Hi Debian has a package for chan_oh323 (the original, external h323). It is not maintaind for quite some time AFAIK and also AFAIK offers no real atvantages over chan_h323. So I'd like to remove it. Before I do that, I have some questions, as I'm not familiar with H.323 channels: 1. Are there any useful features oh323 supports that h323 doesn't? That the version of h323 in 1.4.21 doesn't? That the version in 1.6.1 doesn't? 2. How do I convert the configuration file of chan_oh323 to chan_h323? Just copy over oh323.conf to h323.conf? Any incompatibilities? Any gotchas? 3. Or is it still actually maintained anywhere? -- 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] Weird sip problem
*bump* and more information: I packet sniffed the server during the attempt to call the phone and * never sends a packet to the phone before generating the status is 'UNKNOWN' message. I assume this means that * somehow knows the phone is unavailable, which doesn't make sense to me sense sip show peers shows the phone as OK. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Friday, March 27, 2009 12:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird sip problem I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 ___ -- 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] Weird sip problem
What does the SIP debug say when you attempt to dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Saturday, March 28, 2009 4:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem *bump* and more information: I packet sniffed the server during the attempt to call the phone and * never sends a packet to the phone before generating the status is 'UNKNOWN' message. I assume this means that * somehow knows the phone is unavailable, which doesn't make sense to me sense sip show peers shows the phone as OK. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Friday, March 27, 2009 12:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird sip problem I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.31/2028 - Release Date: 03/28/09 07:16:00 ___ -- 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] callpickup not working
hi folks, Im pretty sure this has been covered before but I just wasnt able to find any answer. Im having troubles with the call pickup feature, is just not working for me. whenever I press *8 or 200 or anyother. nothing happens and sometimes I also get nothing to pickup. I have read this might be a bug although I havent found any patch for it. does anyone have any ideas? Im using BSD 7.1 with * 1.4.6 thanks in advance. --zvonimir ___ -- 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] callpickup not working
On Sat, Mar 28, 2009 at 11:25 PM, Zvonimir Mileta zmil...@hotmail.com wrote: hi folks, Im pretty sure this has been covered before but I just wasnt able to find any answer. Im having troubles with the call pickup feature, is just not working for me. whenever I press *8 or 200 or anyother. nothing happens and sometimes I also get nothing to pickup. I have read this might be a bug although I havent found any patch for it. does anyone have any ideas? Im using BSD 7.1 with * 1.4.6 thanks in advance. --zvonimir ___ -- 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 call pickup didn't work in 1.4.23. It did in 1.4.22, and supposedly works in 1.4.24. But 1.4.6? That's a while ago! sean ___ -- 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
