Re: [asterisk-users] Voice quality assessment in Asterisk
Those boxes run around $50k USD, I've only seen them once back in the late 1990s. At work for customer consulting we have very expensive site licenses for Prognosis IPT Assessor which generate great looking pdf reports. We also use Cisco IOS IP SLA however it doesn't have a reporting mechanism. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bert Van Kets Sent: Friday, October 08, 2010 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice quality assessment in Asterisk The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score http://voip.about.com/od/voipbasics/a/MOS.htm Bert On 08/10/2010 11:12, Sevana Oy wrote: Hi, How do you typically test voice quality in Asterisk? For example if you like to do load testing, or monitor voice quality and get notified if certain calls had bad quality for proactive maintenance? Thank you! Best Regards, Sevana Oy http://www.sevana.fi - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- 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] Opensource Speech recognition for Asterisk
I'm not aware of an open source speech product. Some great examples where speech recognition works well are 1-800-USA-RAIL, Microsoft/Cisco corporate directory 425-882-8080 you can say the employees name and be connected and those works great, 1-800-Goog-411 also works well. Windows 7 Speech Recognition, Dragon Natually Speaking work pretty good. Vonage does a good enough job of sending my home voicemails to my email in Speech to Text, I use this app daily, rarely having to listen to actual voicemails. What Speech-Text doesn't convey is anger/happiness, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Sunday, August 22, 2010 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Opensource Speech recognition for Asterisk On Saturday 21 August 2010 17:21:30 Zeeshan Zakaria wrote: I yet have to see ANY working speech recognition software, free or not. This technology is nothing more than a joke so far, not practical at any level. As for free, there is nothing decent. Actually, speech recognition works fine across the board AS LONG AS you use a limited grammar set. It's the arbitrary language speech recognition that needs to be trained to a particular voice. However, arbitrary language isn't normally a common case for IVR systems, which need a limited set of responses in order to decide the proper branch in a decision tree. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] T.38 fax between ATA's and Asterisk and Cisco PGW 2200
WireShark does a good job showing the T38 communication. Most products you can also set packet redundancy to send 2 packets. Your setup was T.38 ATA to T.38 Gateway with PRI/ANALOG/PSTN/G.711. I've heard various problems with SIP/PSTN and faxing, around jitter/packet loss and it's not supported by Verizon SIP and others. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of P Z Sent: Thursday, July 29, 2010 7:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200 To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax messages from all ATA's to the PSTN (so ATA - Asterisk - PGW - PSTN) failed with a variety of error messages so i tested the different steps one by one. ATA's - Asterisk ReceiveFax: So far i have only succeeded in sending fax messages from the Fritz!Box 7170 to both Asterisk configurations using the ReceiveFax application. Sending fax messages from the other ATA's to Asterisk using the ReceiveFax application failed. ATA's - PGW: To exclude Asterisk i have connected the ATA's directly to the PGW; no success either. Asterisk - PGW: To exclude the ATA's i used the Asterisk SendFax application to send a TIFF file to a landline each time with a different fax machine connected to it. Results: Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 : asterisk[1367]: WARNING[18591]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=19: Received other than DIS while waiting for DIS. asterisk[1367]: WARNING[18591]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18906]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=20: Received no response to DCS or TCF. asterisk[1367]: WARNING[18906]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18986]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. asterisk[1367]: WARNING[18986]: app_fax.c:817 in transmit: Transmission error Asterisk 1.6.2.6 with Fax for Asterisk : asterisk[7092]: WARNING[3167]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f49' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3167]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f49' in T.38 mode asterisk[7092]: VERBOSE[3226]: -- FAX handle 0: [ 028.000627 ], entering CLOSING state asterisk[7092]: VERBOSE[3225]: -- Channel 'SIP/out.to.pgw-000b3f72' FAX session '11' is complete, result: 'FAILED' (FAX_FAILURE_TRAINING), error: '3RD_FRM_CHECK_ERROR', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' asterisk[7092]: WARNING[3272]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f8b' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3272]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f8b' in T.38 mode My questions: - Does anyone have experience with T.38 fax with a setup like this: ATA - Asterisk - PGW - PSTN? - Does anyone have experience in connecting Asterisk to a Cisco PGW 2200 + AS5400XM? - Are there any tools to debug T.38 traffic? Thanks! - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- 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] ringback tone after MOH, before queue member bridged
I normally work with other 3rd party IVRs, usually once the Agent is Reserved we signal the phone system to play Music on Hold while it's ringing the Agent. The trick here is to replace the Music on Hold with a fake ring file. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ad...@3a.hu Sent: Friday, July 23, 2010 3:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ringback tone after MOH, before queue member bridged Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R — stops moh and rings once an agent is ringing (Asterisk Trunk) (in queue's optinal parameters). Could someone please explain this line to me? I've set this option, i have a softphone and an ATA registered to *, pure SIP, nothing more. It's not working, either i'm using the r option, which disables MOH and just rings, or i'm using R which gives me MOH but no ringing. It's nothing major, it just would be nice to have. thanks adam -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] Complex Dialplan Help Needed
I think you need to ask your SIP provider about Redirecting Header, ask what they support and how-to. I work more with Cisco CallManager and SIP Rediversion Header is new in CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco Mobility/Single Number Reach, providers usually won't accept a Calling Party Number that isn't in your range, some will. http://www.voip-info.org/wiki/view/RDNIS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh Sent: Monday, July 12, 2010 6:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Complex Dialplan Help Needed Hello all, I have a project which requires me to rout calls from ten blocks of sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers from the blocks of numbers from the SS7 gateway, not the CLI of the originating caller. The Asterisk is behind a firewall with NAT setup. The traffic is one way only. Calls going to the switch goes to Asterisk, Asterisk accepts the call, looks at the CLI from the line (not the caller), routs the call to its assigned outside number through the primary trunk. If primary trunk is unavailable, trixbox will then rout the call to the spare trunks on the list. Hope anyone who has setup this before could give me some good tips on how to set this up. Geoffrey -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] Software for my laptop to send Fax via H.323 ?
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323. Trying to find a way I could use my laptop to send a fax over H323 to the BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't the router with me! - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- 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] Phones won't stop ringing
I'm experiencing runaway ringing too, can we make this a class action against someone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Brower Sent: Wednesday, March 10, 2010 10:20 PM To: Chris Owen Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Phones won't stop ringing Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] Faxing over Carrier SIP trunk/g711 ?
Anyone have a customer sending/receiving multi-page faxes over Verizon Business SIP trunk/g711 ? Verizon Business indicates they don't support it, and I have 2 recent customers that it doesn't work for, and 1 current large customer telling me he's going to make it work grin. The issues is the latency/jitter on fax/g711 over Verizon Business seems to spit out only 11 pages of a 15 page fax. Anyone having faxing over PSTN SIP over G711 that is working? Any advice? - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calling Number Verification Number? for BellSouth/ATT
BellSouth (now ATT) has a number you can dial and it will play back voice prompts with your calling number? It's used by their techs with a buttset in identifying analog 1FB lines... Eg Dial 704-210-3233, it answers Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from 704-559-2122 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Calling Number Verification Number? forBellSouth/ATT
To clarify the question is what is the number for ATT Calling Number Verification? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Monday, June 29, 2009 7:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calling Number Verification Number? forBellSouth/ATT BellSouth (now ATT) has a number you can dial and it will play back voice prompts with your calling number? It's used by their techs with a buttset in identifying analog 1FB lines... Eg Dial 704-210-3233, it answers Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from 704-559-2122 Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] 400 calls at g711 how much cpu power
Is this Project Eagle Eye ? Call every phone at once to tell them about H1N1 in their neighborhood From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel Business Sent: Monday, June 22, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power Lol , simply lol, don't forget the super duper, top secret patch ,everyone is hiding from you that makes asterisk able to do 4000 calls on a p3, PS. don't tell anyone i said this . But yeah , since you need to blast 500 calls+, you should be aware that normal blasting even 4 seconds audio will run you quite a bit of money, 20 seconds * 400 channels = 8000 seconds every 20 seconds, +- prep times... Or 133 minutes every 20 secs... 399 minutes every minute.. @ 0.015 let's say 400 * 0.015 is 6$ a minute, $360 an hour, 3600$ a day, and ill let you do the weekly fees Now, that's starting to be expensive for a pet project ;) if not and gov related, then ill just pass the remarks.. I always knew there's money in fear, but broadcasting it could be worth it too ;) Can't wait for the day when we get voice calls about buying water in bulk and storing crackers. Anyhow let me know how you manage to do 400 calls on asterisk with or without transcoding From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erick Perez Sent: June-20-09 9:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power I am fairly certain he was simply reporting the results (for posterity) of the event having already happened. Good to know (I guess?) that such small hardware can acheive the performance that was squeezed out of it. Impressive. All THAT said, I am unconvinced that there was no sales effort involved in sending out millions of unsolicited calls. Claim if you like that this was some public information event (which you fail to expand much upon) and convict me of mistrust, but who would have paid for such a thing. TV ads, radio spots, billboards, etc., are much more effective for public information. Unsolicited calls on that order mean only one thing to me - SPAM. So what wonderful product were you informing the public about with regard to the looming threat of illness? Jeff, indeed i was posting for posterity. Maybe someone will benefit in an outbound-only scenario that he/she will not need a supercomputer to pump a 20sec audio clip. Again, this was a public service. And indeed TV and radio was used. Unless you live in a bubble, you may have heard about AH1N1 virus. Which unfortunately hit us (Panama, Republic of Panama, Central America) very hard. I foud very repetitive to tell in my posts that i am from panama, central america, blah,blah blah. Anyways, a quick google search of this forum will also revealed that i am kind of a regular poster and even my cellphone is listed here (Jon Pounder, my cellphone is +507 6675 5083 in case YOU want to sell me a car loan, i dont mind getting a call. Im a IT consultant and i have a chargeback line. Please call me as many times as you want...please do so between 10pm and 6am where my chargeback is the most expensive). Guys, Grow up! Next time someone needs to learn mouth-to-mouth and CPR lessons, please DONT teach him. Because, following your inmature way of thinking, the person who wants to learn CPR may as well be looking for information to learn how to suffocate people. Next time your son wants to know how gasoline works or how is being produced. Please keep your familiy in ignorance. You may be training the next crazy person who will burn things all around the world. But, you wont do that, do you? Again, I always tell my familiy that keeping others in ignorance is bad. but sometimes it must be done for the sake of a greater good, and my comment is always followed with good and sound examples (atomic technology, viruses, etc). But I forgot that Asterisk, the phone lines and a calling system is the way the world is going to be dominated by the martians. So the secret about phone system calculations must be keept in Area 51. Now I understand Kevin Mitnick. Cheers to all. Bye. Erick Perez Cel +(507) 6675-5083 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received
Re: [asterisk-users] Limit transfers
What is to stop anyone from dialing international at any time, regardless if he bridges someone else on? Usually we implement Force Authorization Codes (When dialing out after dialing you have to enter a code) to track all Long Distance/International calls. You can then generate bill back reports to departments by code. Some software (ISI InforTel) can trigger an email alert if someone spends more than $100USD in a day based upon CDR records, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel A. Veiga Sent: Saturday, June 20, 2009 10:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit transfers I have asterisk installed in a callcenter: 60 DAHDI external lines and 72 SIP extensions, and have a BIG PROBLEM. Image a friend of one of the agents wants to call abroad paying the local fee. He dials to the callcenter, uses DID to get to his friend, asks him to place the call on hold and dial abroad and then hangup to bridge the call!!! I discovered this by chance when one of the agents, after answering an external call, tryed to hangup and place a new call. He really pressed the hook for a few milliseconds and was interpreted as a flash. After finishing the call he hanged, bridging the first call with the second. I cannot disable call forwarding, as the agents need it under certain circumstances. What I need to do is disable bridging two external lines. Has anybody faced this problem? Thanks, Daniel ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] AmooCon video recordings online
No divx hd? just kidding OT: Odd how many video/audio standards there are, and the growing issue with them? I recall when you had two choice Windows Media or RealPlayer. Now I have to make 3-4 for everything from DivX to iPod to Walkman. For example my cell phone can't play a H264/AAC due the cpu requirements needsI'm happy to see Windows 7 added H264/AAC natively... Imagine adding 5-6 more audio codecs and having to support them...like we don't have enough already... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 3:49 PM To: Asterisk Users Subject: [asterisk-users] AmooCon video recordings online JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] AmooCon video recordings online
I just wish my HTC Touch Pro cell phone or my PlayStation3 could play .mov -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Friday, June 12, 2009 5:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AmooCon video recordings online Jason Aarons (US) schrieb: OT: Odd how many video/audio standards there are, and the growing issue with them? I recall when you had two choice Windows Media or RealPlayer. There is only one format[1] of choice: .mov :-) It's amazing how formats natively supported by QuickTime play smoothly at high resolution even with 2 virtual machines and all sorts of other stuff running on my MacBook. That's next to impossible with .wmv/.flv videos and Flip2Mac / Perian. divx kinda works but requires an additional player. Apple must have put an incredible amount of work into QuickTime optimizations. Now I have to make 3-4 for everything from DivX to iPod to Walkman. For example my cell phone can't play a H264/AAC due the cpu requirements needsI'm happy to see Windows 7 added H264/AAC natively... [1] Mixing up container formats and codecs here. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] SIP CALL ENCRYPTION
Are conference bridges and other resources going to work with SRTP ? I'm wondering what enabling SRTP will break in Asterisk. It breaks several things in Cisco CallManager. Also wondering what make/model SIP phone you are using for SRTP and what experience other having using that make/model for SRTP? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, May 28, 2009 3:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP CALL ENCRYPTION On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote: Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers I have reviewed and shall soon try: http://www.voip-info.org/wiki/view/Asterisk+SRTP This technically isn't SIP encryption. It encrypts the RTP streams. Though this is probably what you're really after. This still won't e.g. encrypt the dialed number. -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] Fax Machines across carrier SIP trunk? General recommendation?
Customer has a Verizon Business SIP trunk, I'm still used to PRI T1 myself for local service. The fax machines are having some issues (I can use analog phone to call out fine) and I'm checking on modem passthrough with Verizon, but wonder if any else is using Verizon Business for SIP trunk and what your faxing milage was? Did they support G711 and modem-passthough, etc? Also checking QoS, etc. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] DTMF
Is this inbound calls to your automated attendant? Or Outbound calls to say a bank ivr out in the pstn? What direction? What is your interface/carrier? T1, SIP, H32? And what method are you using for DTMF? Eg inband, out of band, what rfc, etc? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Friday, May 22, 2009 3:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] DTMF
Then if it's a IP interface (SIP, etc) have you tried a sniffer trace (wireshark, etc) to verify the packets are being sent correctly to carrier? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, May 22, 2009 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF Is this inbound calls to your automated attendant? Or Outbound calls to say a bank ivr out in the pstn? What direction? What is your interface/carrier? T1, SIP, H32? And what method are you using for DTMF? Eg inband, out of band, what rfc, etc? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Friday, May 22, 2009 3:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Alternative to Adobe Audition 3 for G723 G711 uLaw ? (old Cool Edit Pro)
Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD) can do it -jason - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Hang at 5:34 pm EST
If you set the system clock ahead does problem follow the clock? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May 19, 2009 6:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hang at 5:34 pm EST Some at 5:34 pm EST DAILY, all my call get disconnect. I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its same. I tried changing VOIP provider I tried changing Internet Provider..But no help.. What could be the reason ? Here are my enties of crontab : ### recording mixing/compressing/ftping scripts 0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl #0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl 1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --GSM #2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --GSM ### keepalive script for astguiclient processes * * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl ### kill Hangup script for Asterisk updaters * * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl ### updater for voicemail * * * * * /usr/share/astguiclient/AST_vm_update.pl ### updater for conference validator * * * * * /usr/share/astguiclient/AST_conf_update.pl ### flush queue DB table every hour for entries older than 1 hour 11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q ### fix the vicidial_agent_log once every hour 33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl ### updater for VICIDIAL hopper * * * * * /usr/share/astguiclient/AST_VDhopper.pl -q ### adjust the GMT offset for the leads in the vicidial_list table 1 1,7 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug --postal-code-gmt ### reset several temporary-info tables in the database 2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl ### optimize the database tables within the asterisk database 3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl ## adjust time on the server with ntp 30 * * * * /usr/sbin/ntpdate -u 18.145.0.30 2/dev/null 12 ### remove old recordings more than 7 days old #24 0 * * * /usr/bin/find /var/spool/asterisk/monitor -maxdepth 2 -type f -mtime +7 -print | xargs rm -f ### remove old vicidial logs and asterisk logs more than 2 days old 28 0 * * * /usr/bin/find /var/log/astguiclient -maxdepth 1 -type f -mtime +2 -print | xargs rm -f 29 0 * * * /usr/bin/find /var/log/asterisk -maxdepth 3 -type f -mtime +2 -print | xargs rm -f 39 10 * * * /etc/webmin/cron/tempdelete.pl - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] what can we do with lost voice packet on acongestioned VPN?
I wonder if using the Internet Low Bit Rate Codec or iLBC would work better. G711/G279/GSM all suffer when too many packets are lost. You would then need to transcode to G711, etc -jason http://en.wikipedia.org/wiki/Ilbc -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nik600 Sent: Sunday, April 05, 2009 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] what can we do with lost voice packet on acongestioned VPN? Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing is that on the same network there is an H323 trunk from an Alcatel and a Cisco CCM (using g711 codec) and in that case the voice isn't so bad! i've tried to enable jitterbuffer but i can't notice some difference. Is there something else that i can do? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] [Zaptel] Why no driver for PCI voice modems?
I don't think a off the shelf modem has the necessary DSPs to convert voice to codecthat is why a Voice Gateway/Analog Telephony Adapter or FXO/FXS cards exist instead of modem having a second life. I do recall a few that worked as a answering machine allowing your home computer to answer calls ,etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm Sent: Wednesday, April 01, 2009 1:06 PM To: Asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems? If half-duplex audio is good enough for you, sure. You've lost me there. I am not aware of a modem that is for sale today that is half duplex. (OK some support a couple of minor half duplex modes). All state of the art modem protocols send and receive simultaneously using the full 300 - 3000 Hz bandwidth in both directions with adaptive equalization and echo cancellation to make it work, which is pretty much what a voice circuit need. There are two differences: 1) The response and quality of a current modem must be considerably higher than what is needed for voice use or it would never achieve the throughput expected of it, and 2) the adaptive equalization algorithm is designed around modem specific techniques. The latter is (especially for a softmodem) a software issue, not a hardware limitation. Only a fraction of the hardware available is actually capable of full duplex audio. Absolutely not the case. Particularly the softmodems (the most inexpensive) contain little else than what is required for placing and answering full duplex audio calls. Everything else is in the driver. The OP is 100% correct, that they would be an excellent candidate for FXO use in low volume applications. What it really comes down to is a value proposition: Quite true. This is the real issue. As mentioned, these drivers require considerable skill and knowledge to write. While there is no doubt that the result would be very cost effective, the business model is lacking. The modem manufacturer is going to see the potential market for this as somewhere down in the noise compared to their normal modem sales, so isn't inclined to invest. A third party developer with the skills would have a difficult time recouping development costs (let alone any profit) because they don't control the hardware, and therefore have no leverage. A user with enough volume to justify paying for the development (or doing it if they had the skill) probably has enough volume to use T1s instead. If everyone that could benefit from using a modem card were to pitch in $10 towards the development, it would probably be quite possible. But how to make that happen? Wilton - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Problem with Verizon Wireless
Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Problem with Verizon Wireless
Is the feature you are implementing Single Number Reach? They dial a number and you call another number (Verizon Cell Phone) trying to connect them to the user? But the problem is Verizon answers with the silly out of reach message? I've never seen where the PSTN carrier lets you re-direct the call to the cell phone without your Single Number Reach PBX holding/hairpinning the call. I'm more old school PBX than SIP expert and suspect this can be done in the SIP cloud. I suspect services like Vonage Ring Lists don't hairpin calls! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Monday, March 16, 2009 8:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with Verizon Wireless On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. 4 channels? Could you count them for me please? I'm just getting started and working my way up from the simplest configurations. I may not have the jargon right right. I was expecting that I could eventually configure things so that I could hand off the calls so that once the Asterisk box got a connection between the DID provider originating the call and whatever/whoever is terminating the call (SIP device, or SIP service provider) the Asterisk box could then drop out of the connection and let the originator talk directly to the terminator. Is this an unrealistic assumption. Ah, I see one disconnect. I think you are assuming T1 or better connections to the PSTN where you are originating and terminating the calls yourself and I'm using SIP service providers to do all the origination and termination. I'm connecting a bunch of home offices scattered around the country and do not have enough lines in any city to justify originating or terminating my own PSTN calls. Maybe just one PSTN line per DSL connection to avoid paying a sip provider to terminate some local calls, and supporting some backup functionality, if the Asterix box has crashed, but it will be a while before things get that complicated. ___ From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- 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 Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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
Re: [asterisk-users] DID's in a specific rate center
Any idea what legal statues setting caller-id fraudulently falls under? Is there a federal law you can reference? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber Sent: Wednesday, February 25, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] Please help test the gender detection moduleat 575-613-4392
After helping out it seems I've been determined a female(wrongly). It was disappointing and I'm considering a visit to the Dr Phil Show to work out my anger From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Asterisk Sent: Tuesday, February 17, 2009 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: nt_jnew...@yahoo.com Subject: Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392 That's funny. The way I have it phrased, when I called I started talking to it as well! I have some code for short list voice recognition and thought about detecting yes and no in there, but I ran out of time...and the prompts were already recorded. Thank you everyone for helping test the module. There have been 200+ calls from users on the list and they are still coming in. We're getting about 65%-70% success rate. My target is 80%-85% in random sampling and 90%-95% in controlled settings. Update: I'm adjusting the detection ratios tomorrow, so that should improve general detection results based on the received data. I'm implementing filters to remove the background noise. I'd guess that 5% of those testing are trying to fool the system for fun, in one way or another. When the user is unaware of sampling, the results are slightly higher. My greeting suggests a less masculine phrase, but with a male voice. I suspect this throws off both genders' recordings. I probably should have had testers say their own names, since testers rarely divert on that. From: Gondar Monn gonda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 16, 2009 9:19:20 PM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 Looks like my provider is not passing dtmf correctly .. Had a serious laugh, system kept asking me if I was ready., ended up finding myself talking to the IVR . - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] life safety system and VOIP
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] life safety system and VOIP
In Florida some new subdivision developers have sold the phone/cable/internet rights to a provider. They run fiber to each house and then have the uplink to provider which isn't a traditional telco. You can't get another provider as satellite dishes are limited in covenants and restrictions (CCR). I guess you could get GSM or CDMA service from cell provider or WiMax/LTE. It provides an upfront funding to developer for sewer/water costs. I'd be curios what battery life they have. I know the FCC mandated cell towers have more battery life after Hurricane Katrina wiped out communications in New Orleans for months. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonn Taylor Sent: Tuesday, February 17, 2009 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] life safety system and VOIP Jon Pounder wrote: Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net mailto:j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co's ATM network or the cable companies network (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. This is true, that is why most fire panels have to have 2 phone lines. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter
Re: [asterisk-users] 32 bit server is ok?
The Intel 80386 used 32-bit architecture in 1987...might want to specify make/modelI'm not sure you want to run * on a old Tandy or Packard Bell -jason From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: Thursday, January 29, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 32 bit server is ok? hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Newbie in Cisco Phone
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/p s8759/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying... So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Sip Node w/ 4 wire audio AT command set callsupervision
Google Multitech CallFinder GSM out of Minnesota if you want a common off the shelf product. GSMA.org was using their product with FXO/FXS for backup purposes. I recall they have a GSM to FXO/FXS, and I thought they had GSM to H323. I also found a European company that made high end (24+) GSM banks to FXO/FXS but can't recall the name. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Bean Sent: Saturday, December 06, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sip Node w/ 4 wire audio AT command set callsupervision I have several discontinued Sierra Wireless MP775 mobile GSM/EDGE radio modems. These devices were originally installed in emergency vehicles to provide data and voice access along with GPS reporting. They have external RF connections for GSM/EDGE and GPS signals. The baseband side includes four wire audio, RS-232 serial, USB and some parallel digital I/O for panic switches etc. GSM voice connections are established by sending AT commands via the RS-232 serial or USB ports and utilizing a handset connected to the four-wire audio jack. Here is a link to the Sierra Wireless web pages for the MP775. http://www.sierrawireless.com/support/mp775.aspx My interest comes in finding an inexpensive way to connect an Asterisk PBX or similar system to the PSTN via GSM when POTS and Internet service isn't available or is too costly to connect. In my case, I'm considering building a house at the end of a long unpopulated stretch of dead end road and the cost of trenching and cabling from the nearest telco POP is prohibitive. I would like to find a way to connect these modems to my network so they appear as a SIP FXS device. This would require a device generating/reading AT commands and passing baseband audio on the front end and SIP emulation on the backend. I'm sure there must be a way to do this with a pc but to minimize power consumption; I would prefer to use something like a small single board computer with a Geode processor. The latter, having multiple RS-232 and audio ports, ought to be capable of handling at least two of the MP775's. Has anyone seen any hardware, software or combination that would allow me to accomplish such an interface? Regards, George - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call
Microsoft doesn't make a native SIP client in Windows Mobile you can use for a phone call. Do you mean Windows Live Messenger? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Wednesday, December 03, 2008 9:15 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call I'm using the Wm6 built in client. (Enabled via CAB file to add-back files removed from ROM) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta Sent: December 3, 2008 8:42 PM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call What sip client are you using on WM6 side ? - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Large Asterisk installations (~10, 000 extensions), preferably at universities
Just switching from Nortel to something else may not eliminate hardware/software failures, or prevent those without experience from pushing the enter key at the wrong time. You have to consider the two professionals actually cost considerably more than just salary, due to taxes, 401k, benefits and after they gain knowledge of your Asterisk will they go to a VAR to make big money? It's a catch-22. I would start with a small pilot of anything (Nortel VoIP, Cisco VoIP, Asterisk) and use it to build your knowledge/experience and then make an informed decision versus a rush decision based upon how last week went with the legacy equipment. You might find you need to plan for PoE long term, or have network issues, etc, etc. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine Sent: Friday, November 21, 2008 6:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Large Asterisk installarions (~10,000 extensions), preferably at universities Thanks to everyone who replies so far! We have Nortel PBX'es with a support contract from one of the local VARs (Nortel does not give direct support here). In the last two weeks we had one of our exchanges down for three half days; one was after a failure, and the other two were when the technician came to fix remainders of the original problem and just did a 5 seconds restart which won't even cut calls. Yeh, the 5 seconds took 6-7 hours... BTW, they still do not know what was the original problem. So, why won't we save the big bucks we pay them, hire two professionals (who cost less) and support an open source code by ourselves? This way we depend on ourselves only. Thanks, __Yehavi: 2008/11/21 Grygoriy Dobrovolskyy [EMAIL PROTECTED] 2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Asterisk 1.4 or 1.6 ???
I would stick with 1.4 in production, how mad would you be if I gave you a cell phone with new code and it didn’t work? Would you throw your cell phone at me if it cut us off during phone calls from a bug? Some people are ok with trying new stuff, others it costs money when they lose business due to their phone system not working. I’ve noticed you can mess up computers, but phones get people mad when they don’t work. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Facultad Sent: Monday, October 06, 2008 3:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 or 1.6 ??? Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6. My scenario is: SIP server with 100-150 SIP users, voice mail and maybe IVR. I will use GSM audio codec. Maybe in the future I'll connect a E1 line to the PSTN. What Asterisk version is better to me and why ??? Thank you. A.F. Yahoo! Cocina Recetas prácticas y comida saludable Visitá http://ar.mujer.yahoo.com/cocina/ - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is?
A lot of places you still can't get GSM in the US.it has improved...but GSM 3G coverage is lacking compared to EVDO/CDMA. Another option is a World Phone that can do all bands. My story; Visiting American lands in Kuala Lumpur and checks phone for messages... Puzzled look appears as it worked at Home, Canada, Mexico, Caribbean... You start to explain about GSM and their eyes open wide as they realize they need a unlocked GSM phone from a electronics shop and SIM chip from some company named Digi sold in 7-Eleven and some scratch off cards for refills using SMS. In reality my roaming fees for Intl are too high, I'll get a pre-paid in-country phone before I get phone bill for Intl roaming. My data connection syncs email all day long. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists) Sent: Thursday, September 25, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OT: Do You Know What the Problem With CDMA is? On September 25, 2008 10:41:45 am Drew Gibson wrote: Once CDMA has gone the way of the dodo in North America, I really will miss one of my favourite scenes:- Visiting Brit steps off plane and checks phone for messages... Puzzled look appears as they ask Why doesn't my phone work? It worked fine in France/Italy/Germany/Timbuktu. You start to explain about CDMA and their eyes open wide as they realize they have just stepped back into the cellular stone age... You don't have ATT towers near airports? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with high latency
Jitter is what your describing, it's a bad thing. http://en.wikipedia.org/wiki/Jitter While VoIP may work (third party 128ms echo cancellers, etc) most support organization won't go outside ITU-T G.114 recommendations. I've done Cisco 7940 phones deployed in the Gulf of Mexico on a oil platform using Callmanager based in US in 2003. The company controlled the satellite and prioritized voice, ping was 600ms. Worked well except the local calls were to Venezula which was too expensive per minute from US. Two phones ran up more than $1000US in 30 days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Moore Sent: Tuesday, July 22, 2008 11:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] issue with high latency Not true. Voip is done over satellite every day and those ping times are at least 540 and upwards of in the 700's depending on the technology used. The key here is keeping the latency stable. If the packet flow fluctuates too much in latency this is when a problem arises. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Tuesday, July 22, 2008 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with high latency On 22 Jul 2008, at 14:36, Nhadie wrote: Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 Never going to work with that latency. I would say anything over 150 is probably pushing it. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
I haven't used Asterisk Voicemail but here are Unity Unified Messaging (for Exchange) 5.x/7.x features, in short I think you need to be a Callmanager/Exchange Server shop with heavy integration with ActiveSync/Direct Push/Outlook 2007/OCS2007. The company that created Unity (Active Voice) was a bunch of ex-microsoft guys. If you are not a enterprise/campus or prefer IMAP/SMTP then I don't think you would see any benefits or ROI. I don't think just hanging Unity Voicemail Only off a Asterisk box would be of much value. I like AVST CallXpress http://www.avst.com/products/callxpressMessaging/ for smaller customers. Unity Unified Messaging (for Exchange) 5.x/7.x Using Exchange Administrator it reads/writes directly to Exchange Message Store (not IMAP or SMTP) Phone View (listen to message as callers leave them, control message on Cisco 79xx phone LCD screen) Windows Mobile/Blackberry intergration (has Blackberry plug-in) Single number for fax/T38 Speech Connect (reply to voicemails via Speech to Text or have them read Text to Speech) Mailbox greetings based on Calendar Integration Unity Digital Networking for multiple sites being able to send each other messages There are flash videos and datasheets here; http://www.cisco.com/en/US/products/sw/voicesw/ps2237/index.html Now I will go read the Wiki and see how Asterik handles Voicemail Note there are several version of Unity (Unity Unified Messaging (with Echange), Unity Unified Messaging (with Domino), Unity Connection 2.x, Unity Unified Express). I choose the version with the most integration with Exchange to discuss here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Tuesday, July 22, 2008 3:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco vs Asterisk It's amazing... the man starts the thread with a simple question: Can anybody tell him if Asterisk can do the same things that the Cisco Unity Server can do?, if it can do some better, some the same, and/or some worse, can someone indicate which ones? Also, can Asterisk complement the Cisco call manager functionalities?... I wish I knew the answers, and I am myself interested in the educated straight opinions of some of the members of this forum. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, July 22, 2008 2:15 PM To: Asterisk Users Subject: Re: [asterisk-users] Cisco vs Asterisk Alex Balashov schrieb: The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Was it like watching a 106' plasma at 1080p for the first time? grin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, July 17, 2008 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality Steve Underwood wrote: You might think a standard phone plugged into an adaptor, like a Magic-jack, would be limited to narrow band voice, as that is all the phone was designed for. It turns out most phones only aggressively filter at the low end of the band. They let a lot of energy above 4kHz through, and they do generally sound better through a wideband codec. I found it quite interesting the first time I used a Polycom IP650 (the first wideband capable hardphone to arrive at Digium) that the voice quality was much improved even using G.711! Presumably this is due to using higher quality speakers, mics and other audio bits required to provide wideband audio quality, but it would appear from this (exhaustive) sample set that in fact many phones (even from high end manufacturers) don't even provide the maximum audio quality achievable with G.711. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack and Skype call quality
My understanding is Skype's secret is using the iLBC codec, which Cisco has also licensed for their 79X2 models as well. I travel and lot and in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator will fail the Skype client will work. The iLBC codec can really handle packet loss. Skype High Quality Video with the Logitech Orbit AF on both ends is awesome. I got my family a set for Fathers day. Just amazing video quality. Uses a On2 VP-7 codec that has much lower cpu and other benefits over h.264. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Saturday, July 12, 2008 3:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality Tzafrir Cohen wrote: On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote: C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS I don't know what Magic-jack does (I've never actually seen one), but I know the key thing about Skype that impresses people - its wideband voice codec. A lot of people poo-poo the idea that wideband voice has value in a phone call. They are either close to deaf, or have never tried it. Clarity is profoundly improved. Skype seems to use various tricks to keep the packet flow smooth, but its wideband that makes it sound better than the PSTN. You might think a standard phone plugged into an adaptor, like a Magic-jack, would be limited to narrow band voice, as that is all the phone was designed for. It turns out most phones only aggressively filter at the low end of the band. They let a lot of energy above 4kHz through, and they do generally sound better through a wideband codec. Many modern line interface chips are actually capable of running in a 16k samples/second mode, even though most are programmed for 8k samples/second. I think the ones on the TDM400P type cards can. Some from Silicon Labs certainly can, and chips from Zarlink and others can. The DAA in those cards can work in 16Hz. So they can send higher quality samples to the telco. Provided Zaptel supports it. But then again, it will get lost as soon as it gets converted to digital at the telco, right? I guess I wasn't clear. What I said was only useful for a SLIC to phone connection. It won't be of any benefit for a DAA to PSTN exchange connection, for the reason you state. Anyway, the ProSLIC chip does not seem to support it. Silicon Labs make a Wideband ProSLIC, Si 3216, which is, er, wideband. As I said before, Zarlink and other make them too. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
Digital ISDN used Q931 messages. You should get a disconnect message from telco on the d-channel 23. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Monday, July 07, 2008 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] US T1 Hangup Detection We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled it. I simply do not see _anything_ related to caller ID going on. (not major, I am not even sure the phone company has it setup properly so I need to talk to them first, Verizon) 2) Asterisk is not detecting the far end hangup. Through the Adtran it does, but direct digital it does not. I bridged an incoming call to an analog phone and listened as I hung up the far end (cell-call). I hear a audible click, silence, and then after maybe a half second I hear dialtone. I tried turning on hanguponpolarity switch, tried turning it off, tried turning callprogress on and off, still does not detect the hangup. Am I missing something obvious in Asterisk (this is my first digital hookup)? I read somewhere that Asterisk is already suppose to detect dialtone to know that the far-end hungup. Do I need to call my phone company and get details on exactly how they are triggering the hangup, though I would think with digital it just happens). Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GSM Gateway Project
Google multitech CallFinder 100, has both FXO and FXS interface you can connect. Problem is you call out and don't get simulated ring back while the GSM call is being setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 21, 2008 11:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk GSM Gateway Project There is a commercial product that does just that. I cannot reveal the company name since they are clients of mine but they have a BTS, a retractable 15 foot tower, a laptop or small PC running Asterisk and either do VoIP over VSAT or connect via T1/E1. Mostly government work but they are busy and growing very fast. Thanks, Steve Totaro On Sat, Jun 21, 2008 at 5:38 PM, broadband Voice [EMAIL PROTECTED] wrote: I am thinking about using an existing asterisk box and turning it into a gsm gateway. Has anyone tried this before, adding sonme gsm cards and an antenna. Any ideas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax on FXS
While not on the FXS port itself other things to look for; 1) set the fax machine to disable Super G3 (33.3k), try to force it to use G3 (14.4k). 2) set the fax machine to Disable ECM (Error Correction Mode) 3) If PSTN is T1 check span for errors, etc. 4) Some protocols have fax-passthrough and modem-relay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Saturday, June 07, 2008 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax on FXS On June 7, 2008 11:37:20 am bilal ghayyad wrote: Hi List; What configuration needed to let my FXS send and receive FAX? Your probably going to need to give some more details about your setup before anybody can help you... theres really nothing special you need to configure for an FXS port to attach a fax machine to it... keep in mind that faxing over VoIP is extremely tricky at best, but if your entire call path is TDM then you shouldn;t have much of a problem. -- Matt Watson http://www.mattgwatson.ca ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] T-Mobile and WiFi Voip
Will this work backwards? When I'm at home instead of my cell ringing have the home phone ring? Why would anyone give up revenue from minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, October 09, 2007 12:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T-Mobile and WiFi Voip Yep all the carriers are looking to offer 'voip' services sooner rather than later. Basically it uses the wifi point to access the mobile switching network. Cool part is you will soon be answering your Verizon home phone on your cell when you are 'within range' or your home network. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, 9 October 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T-Mobile and WiFi Voip I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] [EUQR] Re: North American voice BRI - Informal survey
Since many CLECs (Competitve Local Exchange Carriers in NA) offer fractional PRI, combined with Internet/Data, I haven't seen any demand for ISDN BRIs for voice or data since early 90s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, July 04, 2007 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [EUQR] Re: [asterisk-users] North American voice BRI - Informal survey Jeff Davis wrote: Jon Pounder wrote: If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. I just got off the phone with my sales rep. It appears I'm the third person today to ask about this. (I wonder why?) Your rep at Sangoma? Or your reseller? The answer is no it will not work in NA. Their reasoning being that with limited resources they went after the biggest market. I get the impression that there are no plans to write a North American driver as the demand seems to be very low. This is a real chicken-and-egg problem. More people would get BRI if there were affordable hardware for it. I would like to see them write a NAm driver for it. To get them to take the chance, there have to be enough people willing to purchase the card to make them consider it seriously. The other option is a bounty or community support to get it done. The hardware already exists. The more people make noise about this, the better the chances of that happening. -Stephen- ___ --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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] TM Malaysia E1 PRI signaling
Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia? What signaling did they provide, framing, formatting? primary-4essLucent 4ESS switch type for the U.S. primary-5essLucent 5ESS switch type for the U.S. primary-dms100 Northern Telecom DMS-100 switch type for the U.S. primary-dpnss DPNSS switch type for Europe primary-net5NET5 switch type for UK, Europe, Asia and Australia primary-ni National ISDN Switch type for the U.S. primary-ntt NTT switch type for Japan primary-qsigQSIG switch type primary-ts014 TS014 switch type for Australia (obsolete) Jason Aarons Consultant http://www.dimensiondata.com/na http://www.dimensiondata.com/na 904-338-3245 cell For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ --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] Asterisk to Cisco's Rescue...again...AuthenticateLD Calls
Glad to hear you had a workaround.
I would suggest re-queing your TAC case, perhaps you got a outsourced or less
experienced engineer at Cisco. Their support has varied depending on which
city/group you get. Some have more experience then others.
While your 2600 from 2001 timeframe it should work, you can't run any of 12.4T
images over the last 3 years without maxing the DRAM/Flash.
I've got 1200+ Forced Authorization Codes with 4.1(3)SR1 using
2811ISRs/VWIC2-1MFT-T1s running 12.4T with both MGCP and H323 gateways across
20 sites with no issues. Could be the old 2600s IOS as you mentioned.
From: [EMAIL PROTECTED] on behalf of JR Richardson
Sent: Wed 2/21/2007 8:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk to Cisco's Rescue...again...AuthenticateLD
Calls
Hi All,
Just wanted to share a story:
We turned up a new customer yesterday evening, typical situation, Cisco 2600
Router with T1 PRI card pointed to the customer's analog PBX with 2 data T1's
linked back to our network. The router PRI was configured as a gateway on our
CCM 4, like we've done numerous times in the past. The customer needed LD
Authorization codes enabled, got the list 400+, and configured them in the CCM,
no problem. We started passing calls, local was fine but the LD would not
work, turned the LD codes off and LD would work. After engaging Cisco TAC, was
informed that LD coded do not work with this type of gateway device.
After strapping on my Asterisk-Orange Super-Engineer Cape and Goggles, I told
my Cisco Guy to prepend all the LD traffic with a 3 digit code and send it to
one of my Cluster Asterisk Servers. I put in a pattern for matching just this
customers LD traffic as so:
exten = _5551NXXNXX,1,Answer
exten = _5551NXXNXX,2,Set(CDR(userfield)=Company LD)
exten = _5551NXXNXX,3,Authenticate(/etc/asterisk/companyld.codes|a)
exten = _5551NXXNXX,4,Goto(ccmtrunkld|${EXTEN:3}|1)
;Answer because the authenticate cmd sends audio back to the caller
;set the CDR userfield to the company name
;Authenticate with file where the LD codes are stored, 'a' option puts the LD
code in the CDR accountcode field
;strip the 555 off and pass the LD call outbound
So now all the LD traffic from this customer can be authenticated from the
codes in the file companyld.codes, the CDR is updated properly for parsing the
LD and generates a nice monthly report for tracking who is using LD for this
customer.
I guess I'm feeling grateful that the Cisco Gateway is passing calls in the
first place, but it would have been nice for the Cisco CM and the Cisco Gateway
to play nice together.
The real hero here is Asterisk, Digium, and the Community that supports it!
Thank you All
JR
JR Richardson
Engineering for the Masses
-
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[asterisk-users] RFC2833 SIP trunks and DTMF
I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC 2833? - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ --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] XO SIP Origination Services
I thought XO was reselling Level 3s (old Genuity assets) network/voip just like Qwest ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 11, 2006 3:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Load balance Asterisk server,when it is a SIP client. I am little confused on load balancing, when asterisk server is also a sip client. Based on these, XO Communications one of the largest US DID Provider, now offer SIP Orignation Services for wholesale. Verizon Communications One of the largest US Teleco, now offer SIP Orignation Services. That means no need for PRI card. So if I take service from them, then my asterisk server will be SIP client. Right? How can I set up my asterisk servers so that the calls originated by XO/Verizon goes to different asterisk servers based on load. Has any one does this and can share that with me. Any idea or hint will be appreciated. Thank you, -Kunal, ___ --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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] Cisco 7960G SIP firmware 8.4
For DND press Call Forward All (CFwdAll softkey) then Messages button on the SCCP version. I havent seen the SIP version of 7961G. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, August 30, 2006 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960G SIP firmware 8.4 Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this. On 8/30/06, Hermann Wecke [EMAIL PROTECTED] wrote: Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager ___ --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 -- Lacy Moore Aspendora, Inc. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] 911 versus 9.911
Is there a FCC or other North America requirement that I provide 911 versus 9.911. I want to require users to dial 9.911 in our office, and remove 911. Are there any statutory requirements or laws about this? User accidentially dial 9 then 1 then another 1 and hangup. Weve educated them to stay on the line and ever hang up, but they hang up anyway, resulting in fines for excess hangups to 911. Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] Does anyone use T.38?
XMedius is a great T.38 fax product, integrate with LDAP/AD/Exchange. Integrates with the PRI card in our Cisco Routers using H.323. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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] Prague PTT?
Is anyone familiar with the Telco in Prague? We have an issue with the connection that will be made from the Telco demark when we do an IPT installation next week. -jason - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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
