Re: [asterisk-users] Queue calls drop to voicemail intermittantly
Are the agents ignoring the calls while their ringing? -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack Sent: Monday, December 17, 2007 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue calls drop to voicemail intermittantly Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? ___ --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] Open CallerID Database?
Hey Guys, I'm glad to see this ignited some discussion. I definitely understand there's some legal implications involved, both on a privacy level, and fraud prevention. Obviously an end-user (ie: the person controlling a listing) has to consent to some sort of release resolving the privacy concerns. I'm somewhat aware of the legal implications involved with storing such personally identifiable information (or whatever the legal term is) and have a concern in making sure such issues are resolved. In reality, how is it efficient for every provider to be running their own database? In my mind, this leaves the horribly evident inaccuracies, and even efficiency issues. Thank God these accuracies aren't integral to the operations of telephony systems. I do understand there is a price to pay for such infrastructure, and I believe that it's obvious the telephony world is riddled with racketeering, price gouging ventures, including companies that charge nearly a $0.01 for a lookup. I realize the following analogy is poor, but in mind this is as close as a internet search engine charging for a basic search query. Infact a basic internet query is much more complex, much more costly (ie: the infrastructure of said systems), and yet self-subsidizing. And to the poster who suggested that I was implying scrapping the results from 411.com, this is definitely not even a remote idea in my mind at all. The basis for my idea was a open, moderated, database that was user controlled and self-subsidized. I know this is way off topic, but I really feel that the telecom industry as a whole, and I'm sure I'm not the only one with this belief, is horribly bloated, running on business models that are clearly 30 years outdated. It is 2007, and with the help of the internet, the exchange of information, these telcos now have real, global competition, and real issues to deal with. Anyways guys, I'm curious to hear your thoughts. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Open CallerID Database?
Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Open CallerID Database?
Hey Shane, The basis of my idea was that it would be user-moderated/generated. A 'owner/operator' of a number, would submit verify their phone number, enter their caller id, and basically be done with it. The logistics of it I don't really think would be that complicated. If a listing needs to be updated they basically go through the same process. Right now, we're using a commonly available script (I can't remember the link off hand) that uses Google, 411.com, etc, to do a lookup and although it works pretty good, it is horribly inaccurate the majority of the time. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Monday, February 19, 2007 12:46 PM To: Robert Norton - SophMedia LLC Subject: Re: [asterisk-users] Open CallerID Database? Robert On the surface, I don't see how you could a db with a very good hit rate without paying for the data. There are thousands and thousdands of database updates every day. Perhaps I am missing your intent here. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. --Shane ___ --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] OT: Quad-band cellphones with wifi stable sipsupport
Hey Tomer, I'm not sure if the Audiovox PPC6700 is quad band, but it does support Wifi and runnings SJPhone great! It is even usable over Sprints EVDO service. On Mon, 15 Jan 2007 08:01:44 +0200, Tomer Horn [EMAIL PROTECTED] wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) Thanks, Tomer. ___ --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 -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Caching Caller Name
Hey Guys, Kind of an off the ball question, but I have a random idea. My voip provider supports caller id lookups, but charges a bit for each lookup. I'm thinking about writing a basic PHP script (through AGI) to cache these lookups for say 90 days. Is there an obvious problem with this? Thanks -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Identifying Queue on Cisco 7960
Hey Guys, I apologize for my ignorance on this one. I've got several 7960s running on Asterisk1.4 with 15 or separate queues and am trying to figure out a way to identify to the 7960s, what queue the incoming call is on? Is this possible at all? Thanks! -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Nat Question
Hey, I've got the same exact problems and ended up just giving up. I'd love to hear a solution. On Fri, 12 Jan 2007 17:55:36 -0300, [EMAIL PROTECTED] wrote: Hello all, iam setting up an asterisk box behind NAT to get SIP calls from outside or internet. In that eschema i can setup SIP calls but, while from the outside nat people can hear me, Im unable to listen anything behind NAT. Out of firewalls settings( I checked this to port fowarding) what can i do to get this working fine?. Thanks G. ___ --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 -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --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] Festival Problems
Hello, Hopefully I'm posting to the correct list, but if not, please shun me ;). I'm running Asterisk 1.4, with Festival 1.4.1. I've got a test extension setup, Festival configured and for some reason, when I dial that extension I get this: [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER See the full debug below: [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:3381 sip_answer: SIP answering channel: SIP/SMEDIA-300-086da000 [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6255 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6003 add_sdp: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: True [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6004 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6155 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6200 add_sdp: Done building SDP. Settling with this capability: 0x10c (ulaw|alaw|g729) [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1767 pbx_extension_helper: Launching 'Festival' [Jan 10 17:16:05] DEBUG[9082]: config.c:733 config_text_file_load: Parsing /usr/local/etc/asterisk/festival.conf [Jan 10 17:16:05] DEBUG[9082]: app_festival.c:370 festival_exec: Text passed to festival server : This is just a test at extension 111 [Jan 10 17:16:05] DEBUG[9082]: app_festival.c:433 festival_exec: Cache file exists, strln=36, strlen=36 [Jan 10 17:16:05] DEBUG[9082]: app_festival.c:435 festival_exec: Size OK [Jan 10 17:16:05] DEBUG[9082]: app_festival.c:452 festival_exec: Reading from cache... [Jan 10 17:16:05] DEBUG[9082]: app_festival.c:473 festival_exec: Passing data to channel... [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER [Jan 10 17:16:05] DEBUG[9082]: pbx.c:2363 __ast_pbx_run: Spawn extension (local-phone,111,2) exited non-zero on 'SIP/SMEDIA-300-086da000' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'x300 4806265449' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '4806265449' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '111' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'local-phone' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/SMEDIA-300-086da000' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Festival' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'This is just a test at extension 111' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-01-10 17:16:05' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-01-10 17:16:05' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2007-01-10 17:16:05' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '0' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '0' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1168474565.2' [Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Jan 10 17:16:05] DEBUG[9082]: channel.c:1558 ast_hangup: Hanging up channel 'SIP/SMEDIA-300-086da000' [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:3233 sip_hangup: Hangup call SIP/SMEDIA-300-086da000, SIP callid [EMAIL PROTECTED]) Any ideas? -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development ___ --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
