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As Kevin Fleming Says “So Long, And Thanks For All The Fish!”, We Say Thank You – And Look To The Future

It’s amazing what you can learn in a few days…

Having just found out that Queen Elizabeth has a great sense of humor, it has now emerged that Kevin Fleming – a man who (both with and without his moustache) has been an amazing contributor and influencer in the Asterisk project is set to move on to a new challenge outside the project – but still within the realms of Open Source.

Kevin has been involved with Asterisk for 7+ years, and has been both a thought leader and a powerful voice in the Asterisk world during that time. I first met Kevin at a TMC event called VoIP Developer in California (old school, well before the days of IT Expo), where he was speaking about Asterisk as well as helping to man the Digium booth at the event.

I’ve also followed Kevin around Berlin looking for great gelato during the AstriCon Europe 2006 tour – and it was well worth it, that man knows his gelato!

I’d like to take this opportunity to say thanks to Kevin for his enormous contribution to the Asterisk Project. Without his efforts, Asterisk would not be the success it is today … Anyway, back to main theme – when someone in a senior role like Kevin moves on, it is important that others are there to pick up his responsibilities and move the project forward.

As it turns out, we’ve already been working on this, and have some very talented people that will be taking up the key responsibilities of the project going forward. Some of them have been involved with Asterisk for several years, and some are recent additions, but together they form a great team to lead Asterisk into the future.



Matt Jordan has assumed the project leader role for Asterisk, and is responsible for managing the releases of Asterisk, as well as all of the development efforts within Digium. Mark Michelson is serving as the Technical Lead for the project, responsible for architecture and design direction. We have also recently created the role of Community Support Manager, which Rusty Newton has filled. Rusty is a long time Digium employee with many years supporting Asterisk and Digium products, and will be the day to day interface for community technical issues.

As you know, I recently joined Digium to look after the interests of the worldwide Open Source Asterisk community and I will therefore also be working alongside the good people identified above, especially Rusty.

So while we wish Kevin all the best as he moves on, we are also confident that the good work he and the rest of the team have done continues to be in the best hands going forwards.

To the future…

David

Digium logo David Duffett Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com · http://asterisk.org

So Long, And Thanks For All The Fish!

I’ve been with Digium for just over seven years, and it’s been an incredible experience that I wouldn’t have traded for anything. When Mark Spencer invited me to visit Digium (and Huntsville) in early 2005, I could not have dreamed that I’d end up working for such an exciting, innovative company, finding a wife, and meeting hundreds of people (many of whom are now friends) around the world. It’s been a time of tremendous personal and career growth, and my wonderful colleagues at Digium and in the Asterisk open source community have been directly responsible for most of that.

Recently, though, I’ve been presented an opportunity to take on a new challenge and this has resulted in my acceptance of a new job, in a new industry. In the middle of September, I’ll start working for Bloomberg, L.P., in the Office of the CTO, helping to lead their nascent open source initiative. I’ll be working to bring the power of open source software, open standards, and community building to the financial market data services industry, where it is sorely needed (and overdue). Michelle and I will be relocating to the greater New York City area, but Michelle will continue in her role as Digium’s in-house counsel. Because of our need to relocate, I’ll only be at Digium until August 8th, although I’ll be in Huntsville until around Labor Day.

This is yet another incredibly exciting, career changing opportunity in my life, and I can’t wait to see what it will bring. I’ll be forever thankful for the opportunity that Digium and the Asterisk community provided me to learn, grow and find the place where my skills and experience are the most valuable (to both myself and my employer).

Digium IP Phone D40 Quality, Very Bad

Hi All;

Really it is miserable.

I bring 8 Digium Phone D40 and I used them with a customer, the voice quality is bad internally (between the extension), there is no clearance at all ! We are hearing the voice like another person.

The used codec is ulaw.

The firmware version is: 1_1_0_0_48178

Even at the web based configuration at the phone it self, I am not able to do reboot (there is no reboot button) and I can do this only from the Phone it self.

From the speaker, the voice is very bad and weak.

I am really feel disappointed why I did not use Polycom.

Can someone help me or advise me what to do in this?
Regards Bilal

Static Noise On Bridged Calls To PSTN, Although The Trunk Line Is Clean On Its Own

I have two setups with SIP hardware phones as extensions and POTS lines as trunks. Internal SIP to SIP calls are crystal clear, but all calls bridged to POTS have a significant amount of static noise. The problem is that if I plug a POTS phone directly into the line, there is almost no static noise – the line is clean. It’s like Asterisk (or the hardware) amplifies the static noise. What I’ve tried so far:

1. Connect Asterisk with a short cable directly into the master phone socket, where it enters the building.
2. One of the lines carries ADSL – so I double filtered it.
3. Tried three different phone sets (one Grandstream, two Cisco models).
4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB adapter as analogue-to-digital interfaces.
5. Reduced the software echo canceller in chan_dahdi.conf to 32 and even
16 – until I could actually start to hear echo. Still no difference.
6. Reduced the rxgain and txgain in chan_dahdi.conf to 0 – but the static noise is still there.
7. Tried different phone cables for the pots line.
8. Tried a different motherboard on the computer with Asterisk and checked there is no IRQ sharing. Tried when there was no other load on the Asterisk computer.
9. Tried Asterisk 1.6, 1.8 and 10

Is there anything else I can do – or should I just give in to the static noise? Is that how other hybrid setups work – do you get static noise on the line – more than if plugged directly? The client is adamant that the noise on the line is too high – by comparison with the quality on mobile phone calls (which are digital, incidentally) – so if I don’t find a solution, I suppose I will just have to rip it all out and let one of the companies with proprietary phone systems install one.

Any hints appreciated.

Sebastian

AGI Not Generating Sip 180/183 Status

hello,

i have strange problem with AGI (asterisk 1.8.10.0)
when i use Dial from dialplan everything is ok when i dial from AGI script there is missing SIP Status 180 ringing and
183 session progress

any ideas?

DIAL without AGI

196.356479 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE
sip:222333444@some.pbx.org, with session description
196.356768 10.0.0.213 -> 10.0.0.193 SIP Status: 401 Unauthorized
196.365709 10.0.0.193 -> 10.0.0.213 SIP Request: ACK
sip:222333444@some.pbx.org
196.370028 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE
sip:222333444@some.pbx.org, with session description
196.370503 10.0.0.213 -> 10.0.0.193 SIP Status: 100 Trying
199.797325 10.0.0.213 -> 10.0.0.193 SIP Status: 180 Ringing
199.797932 10.0.0.213 -> 10.0.0.193 SIP/SDP Status: 183 Session Progress, with session description
199.878441 10.0.0.193 -> 10.0.0.213 RTCP Receiver Report Source description
199.988259 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seqr89, Time171500, Mark
200.004139 10.0.0.213 -> 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, SeqP775, Time(960
200.008118 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seqr90, Time171660
….
201.504218 10.0.0.213 -> 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, SeqP850, Time@960
201.519477 10.0.0.193 -> 10.0.0.213 SIP Request: BYE
sip:222333444@10.0.0.213:5060
201.519611 10.0.0.213 -> 10.0.0.193 SIP Status: 487 Request Terminated
201.519800 10.0.0.213 -> 10.0.0.193 SIP Status: 200 OK
201.528465 10.0.0.193 -> 10.0.0.213 SIP Request: ACK
sip:222333444@some.pbx.org


DIAL from AGI
66.581752 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE
sip:222333444@some.pbx.org, with session description
66.581958 10.0.0.213 -> 10.0.0.193 SIP Status: 401 Unauthorized
66.590738 10.0.0.193 -> 10.0.0.213 SIP Request: ACK
sip:222333444@some.pbx.org
66.595555 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE
sip:222333444@some.pbx.org, with session description
66.596167 10.0.0.213 -> 10.0.0.193 SIP Status: 100 Trying
66.652571 10.0.0.213 -> 10.0.0.193 SIP/SDP Status: 200 OK, with session description
66.676485 10.0.0.193 -> 10.0.0.213 RTCP Receiver Report Source description
66.750371 10.0.0.193 -> 10.0.0.213 SIP Request: ACK
sip:222333444@10.0.0.213:5060
66.844392 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq869, Time20100, Mark
66.854430 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq870, Time20260

69.404625 10.0.0.193 -> 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq998, Time40740
69.516390 10.0.0.193 -> 10.0.0.213 SIP Request: BYE
sip:222333444@10.0.0.213:5060
69.516669 10.0.0.213 -> 10.0.0.193 SIP Status: 200 OK

Asterisk 10.7.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 10.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* — Fix deadlock potential with ast_set_hangupsource() calls.
(Closes issue ASTERISK-19801. Reported by Alec Davis)

* — Fix request routing issue when outboundproxy is used.
(Closes issue ASTERISK-20008. Reported by Marcus Hunger)

* — Set the Caller ID “tag” on peers even if remote party
information is present.
(Closes issue ASTERISK-19859. Reported by Thomas Arimont)

* — Fix NULL pointer segfault in ast_sockaddr_parse()
(Closes issue ASTERISK-20006. Reported by Michael L. Young)

* — Do not perform install on existing directories
(Closes issue ASTERISK-19492. Reported by Karl Fife)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0

Thank you for your continued support of Asterisk!