- VoiceOps - lists.voiceops.org

using agile for service delivery
by acowan＠leonidsystems.com April 13, 2012

April 13, 2012

If you're involved in CPE testing or product/service design, you might be interested in a paper on the application of agile user stories to enterprise telephony: http://www.leonidsystems.com/publications/story-driven-service-design-an-in… If you're not familiar, agile is a development methodology popular out here in Silicon Valley with software companies but you can apply it to a lot of things; service delivery is certainly one of them. We're always learning how to better apply the ideas, so please post here or write to acowan at leonidsystems.com if you're at all inclined. Alex Alex Cowan Leonid Systems 415-483-4529 acowan at leonidsystems.com

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response code mapping to upstream peers
by hiersd＠gmail.com April 11, 2012

April 11, 2012

Hi, Does anyone do any response code mapping on the peer side? You can deny information to attackers by changing you codes from the default (turn a BUSY HERE into a BAD GATEWAY, at the SBC, for instance), and I'm wondering if anyone has found a good reason to do so. Thanks, David

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Playback of a G.722 RTP stream
by abaird＠telesphere.com April 11, 2012

April 11, 2012

Does anyone know of a way to capture and playback a g722 rtp stream? I'd prefer to be able to use wireshark and some sort of decoder tool, much like the decoder I use for g729. In fact the company that makes the g729 decoder (http://www.voiceage.com/freecodecs.php) has a decoder for g722.1 and g722.2 but not g722 itself. Are there any tools (free or otherwise) out there that you've seen that will decode a g722 stream for playback? Thanks ADAM BAIRD | SR. VOICE ENGINEER Anywhere (480) 289-7476 | Fax (480) 289-7477 | abaird at telesphere.com<mailto:abaird at telesphere.com> Telesphere | 9237 E. Via de Ventura, Ste 250 | Scottsdale, AZ 85258<http://maps.google.com/maps?f=q&q=9237%20E.%20Via%20de%20Ventura,%20Ste%202…> [cid:image001.jpg at 01CD17DB.90889800]<http://www.telesphere.com/advanced_features.html> [cid:image002.jpg at 01CD17DB.90889800]<http://www.telesphere.com/> [cid:image003.png at 01CD17DB.90889800]<http://www.facebook.com/telesphere> [cid:image004.png at 01CD17DB.90889800]<http://www.twitter.com/telesphere> [cid:image005.png at 01CD17DB.90889800]<http://www.linkedin.com/companies/telesphere> [cid:image006.png at 01CD17DB.90889800]<http://www.youtube.com/user/telesphere> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. Any views or opinions presented in this email are solely those of the author and do not necessarily represent those of Telesphere.

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Broadworks Shared Call Appearance
by telephonetoughguy83＠gmail.com April 11, 2012

April 11, 2012

On Wed, Apr 11, 2012 at 8:01 AM, Andy Smith <telephonetoughguy83 at gmail.com>wrote: > Thanks to everyone for the info. I was misunderstanding a few things, but > I've got it straightened out now. > > > On Tue, Apr 10, 2012 at 3:59 PM, Michael Bain <mike at mostly-harmless.ca>wrote: > >> You should just need to use the %BWLINEPORT-2%, %BWAUTHPASSWORD-2% >> equivalents for the second line (and %BWLINEPORT-3% for the third, etc). >> The Shared Line tag works the same way. You should be able to use >> %BWSHAREDLINE-1%, %BWSHAREDLINE-2%, .... >> >> Mike >> >> On 10 April 2012 13:33, Andy Smith <telephonetoughguy83 at gmail.com> wrote: >> >>> Does anyone have experience using Broadworks Shared Call Appearance on a >>> Polycom Soundpoint IP phone? I can make it work by manually editing the >>> Polycom config file, but I can't figure out how to automate the process. >>> For example, Line 1 on my phone will be my normal private line. The >>> template config file has Broadworks system variables such as, >>> %BWLINEPORT-1%, %BWAUTHPASSWORD-1%, etc. that automatically populate >>> certain fields. I want Line 2 to be a shared call appearance line. Under >>> "reg.2.type" in the config file, I'm using the system variable >>> %BWSHAREDLINE% to change the line from "private" to "shared", but I can't >>> seem to find any variables for populating the User ID, extension, >>> authentication password, etc. for the second line. How do I populate the >>> correct information in for Line 2? >>> >>> _______________________________________________ >>> VoiceOps mailing list >>> VoiceOps at voiceops.org >>> https://puck.nether.net/mailman/listinfo/voiceops >>> >>> >> >

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Broadworks Shared Call Appearance
by telephonetoughguy83＠gmail.com April 10, 2012

April 10, 2012

Does anyone have experience using Broadworks Shared Call Appearance on a Polycom Soundpoint IP phone? I can make it work by manually editing the Polycom config file, but I can't figure out how to automate the process. For example, Line 1 on my phone will be my normal private line. The template config file has Broadworks system variables such as, %BWLINEPORT-1%, %BWAUTHPASSWORD-1%, etc. that automatically populate certain fields. I want Line 2 to be a shared call appearance line. Under "reg.2.type" in the config file, I'm using the system variable %BWSHAREDLINE% to change the line from "private" to "shared", but I can't seem to find any variables for populating the User ID, extension, authentication password, etc. for the second line. How do I populate the correct information in for Line 2?

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Has anyone used the CISCO 6941 with BroadSoft/ACME
by ssrinivasan＠telesphere.com April 9, 2012

April 9, 2012

Hi, Has anyone used the CISCO 6941 IP phone running the SIP load with BroadSoft/ACME? The data sheet says it does SIP with the CISCO UC platform Thanks, Sanjay www.telesphere.com<http://www.telesphere.com>

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Large LNP Orders - Fee?
by Marty.Sorensen＠adp.com April 6, 2012

April 6, 2012

Does anyone charge any kind of a fee to customers for large ports...i.e. over 1000 Tns? You can respond off list if you wish. Martin Sorensen Director, IPNS ADP Dealer Services 3rd Floor West 2525 SW First Avenue, Portland, OR 97201 Phone: (503) 205-4438, Fax: (503) 402-3277 Email: Marty.Sorensen at adp.com<mailto:Joe_Bell at adp.com> This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

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CLI sim testing
by abalashov＠evaristesys.com April 4, 2012

April 4, 2012

I don't know how big your operation is, but I have difficulty visualising an ITSP whose revenue genuinely supports such an expensive venture for the narrow purpose of catching merely one kind of marginal problem, where the marginal utility of doing so would truly exceed the marginal cost. Perhaps you are the exception. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Colin Brown <zavoid at gmail.com> wrote: >Because "should" and "is" are two different things. I've been burned too many times by "should" differing from what "actually is" so I'm looking to get the actual data delivered locally sent to >My NOC in near real time. > > > > >On Apr 3, 2012, at 10:16 PM, Alex Balashov <abalashov at evaristesys.com> wrote: > >> Is that really worth it? Why not just order a bunch of call forwarding products and/or forwarded DIDs around the world, and forward them back to an incoming number on your side? Route it to some clever program. The caller ID should be passed through end-to-end. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Decatur, GA 30030 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> Colin Brown <zavoid at gmail.com> wrote: >> >>> Does anyone have any experience or products you use that you like for remote international CLI testing? >>> >>> basically I'm looking for a bunch of boxes I can put SIM cards into in various countries around the world that I can route my calls to and verify the CLI being set is being delivered correctly(and get that information back to my severs for reporting) >>> >>> Thanks >>> >>> Colin >>> _______________________________________________ >>> VoiceOps mailing list >>> VoiceOps at voiceops.org >>> https://puck.nether.net/mailman/listinfo/voiceops

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SipX for service providers?
by abalashov＠evaristesys.com April 4, 2012

April 4, 2012

If you have a strong development team, don't leave SEMS (SIP Express Media Server) out of your evaluation. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com PE <peeip989 at gmail.com> wrote: >We use Broadsoft as our primary application server and are considering >developing some point apps around it (IVR, other advanced call control, >etc). Besides looking at some commercial products, we are also considering >open source (we have a strong development team). That said, anyone have >experience with SipX that you can share? How does it compare with Asterisk? >I am mostly interested in learning about how easy/difficult it is to manage >and maintain, as well as info about reliability. > >Thanks in advance. > >_______________________________________________ >VoiceOps mailing list >VoiceOps at voiceops.org >https://puck.nether.net/mailman/listinfo/voiceops

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SipX for service providers?
by peeip989＠gmail.com April 4, 2012

April 4, 2012

We use Broadsoft as our primary application server and are considering developing some point apps around it (IVR, other advanced call control, etc). Besides looking at some commercial products, we are also considering open source (we have a strong development team). That said, anyone have experience with SipX that you can share? How does it compare with Asterisk? I am mostly interested in learning about how easy/difficult it is to manage and maintain, as well as info about reliability. Thanks in advance.

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