Re: [asterisk-users] ZRTP + asterisk and Best Security Practice
14 dec 2007 kl. 11.20 skrev Andres Gomez: Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Work is still in progress and we hope to have something ready for testing after new year's. Are Zphone and ZRTP the future for the Voip Security? There's a lot of progress in other areas too. Also remember that confidentiality of the media stream is only one small piece of the larger VoIP security puzzle. Even if the media is encrypted by ZRTP, signalling might reveal information that you consider private. /O --- * Olle E Johansson - [EMAIL PROTECTED] * Asterisk SIP masterclass, Stockholm, Sweden, Jan 2008 * http://edvina.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
[asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Hi, Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. Just my 2 cents I have more than 70 running servers installed with 1.2, we also built our custom interface around it, our custom linux/asterisk distro has been polished over the years and now finally we are earning the profit of all the work we did in the past. We just decided to open a new project with 1.4, but it will take us more than one year, i think, to release the first usable version. So, in the end, my opinion is that is just a matter of time. Hope it helps, have a nice Christmas everyone! -- I migliori saluti,Scrivi a: Alessio[EMAIL PROTECTED] ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Hello everybody, Since 1.4 release our company installed more then 200 Asterisk servers using Asterisk 1.4 version. At start we had several bugs with SIP channel and CDR handling but starting from 1.4.6 or something it works without problems. We are really happy with 1.4 and thank you for your great job! Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: Saturday, December 15, 2007 12:57 PM To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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
[asterisk-users] OpenVox B800P and asterisk 1.4/ mISDN-1_1_7
Hi i've installed this software: SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh jitterbuffer=4000 jitterbuffer_upper_threshold=0 context=misdn language=en musicclass=maracaibo senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=yes need_more_infos=no method=standard dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no presentation=-1 screen=-1 echocancelwhenbridged=no jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no [TEports] ports=1,2,3,4,5,6,7,8 context=from-pstn msns=* MISDN.CONF When i start asterisk i get tihis warning: ** ASTERISK CLI mISDN_close: fid(19) isize(131072) inbuf(0xb6fac008) irp(0xb6fac008) iend(0xb6fac008) == Parsing '/etc/asterisk/misdn.conf': Found [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:929 _build_general_config: misdn.conf: jitterbuffer=4000 (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:929 _build_general_config: misdn.conf: jitterbuffer_upper_threshold=0 (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:985 _build_port_config: misdn.conf: echocancelwhenbridged=no (section: default) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=3,4,5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=4,5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf : ports=(null) (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. P[ 0] Got: 1 from get_ports P[ 1] this is a unknown port type 0x == Registered channel type 'mISDN' (Channel driver for mISDN Support (Bri/Pri)) == Registered application 'misdn_set_opt' == Registered application 'misdn_facility' == Registered application 'misdn_check_l2l1' P[ 0] -- mISDN Channel Driver Registered -- chan_misdn.so = (Channel driver for mISDN Support (BRI/PRI)) ** ASTERISK CLI and in the kernel prints that in dmesg: * DMESG mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) mISDN_dsp: DSP clocks every 128 samples. This equals 4 jiffies. mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range * DMESG Can you help me to guess the problem? Thanks -- /*/ nik600
[asterisk-users] Open ITU G.107 Implementation to measure voice quality
Hi, Does anybody know where I can find any open source ITU G.107 implementation available? I'm looking a way to measure the voice quality in my projects.. Thanks in Advanced, My Best Regards, Andre Lomonaco ___ --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] ZRTP + asterisk and Best Security Practice
Hi Olle 2007/12/15, Olle E Johansson [EMAIL PROTECTED]: 14 dec 2007 kl. 11.20 skrev Andres Gomez: Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Work is still in progress and we hope to have something ready for testing after new year's. Are Zphone and ZRTP the future for the Voip Security? There's a lot of progress in other areas too. What do you have in mind ? Are you thinking about another way to exchange encryption keys ? Also remember that confidentiality of the media stream is only one small piece of the larger VoIP security puzzle. Even if the media is encrypted by ZRTP, signalling might reveal information that you consider private. /O --- * Olle E Johansson - [EMAIL PROTECTED] * Asterisk SIP masterclass, Stockholm, Sweden, Jan 2008 * http://edvina.net Cheers ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ When Digium starts using 1.4 in ABE then I would consider using it in a production environment. All I ever hear is soon, and I have heard that for months if not the whole year. Until Digium itself is comfortable selling and supporting this version, then neither am I. Thanks, Steve Totaro ___ --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] ZRTP + asterisk and Best Security Practice
15 dec 2007 kl. 14.48 skrev Olivier: Hi Olle 2007/12/15, Olle E Johansson [EMAIL PROTECTED]: 14 dec 2007 kl. 11.20 skrev Andres Gomez: Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Work is still in progress and we hope to have something ready for testing after new year's. Are Zphone and ZRTP the future for the Voip Security? There's a lot of progress in other areas too. What do you have in mind ? Are you thinking about another way to exchange encryption keys ? I was thinking of the efforts of using UDP+DTLS to encrypt UDP signalling and SRTP for media. The key exchange and the identify handling is a large area. /O ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
15 dec 2007 kl. 15.42 skrev Steve Totaro: Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When Digium starts using 1.4 in ABE then I would consider using it in a production environment. All I ever hear is soon, and I have heard that for months if not the whole year. Until Digium itself is comfortable selling and supporting this version, then neither am I. Steve, That's very good feedback. Let's try to find out what's holding them. /O ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
One of the biggest barriers to upgrading are the number of little gotchas in syntax changes that can make an upgrade from 1.2 to 1.4 quite painful. After the pain I went through upgrading to 1.4, I've always been recommending to people to think twice about upgrading if 1.2 does what they require. Many of the changes may have been seen as minor - one or two changes are to be expected, but I ran into at least half a dozen - mostly variable changes if I recall correctly - things such as deprecating CALLERIDNUM in favour of CALLERID(num). Some of the breakage was minor (e.g. loss of caller ID processing) but some of them resulted in calls being dropped in unpredictable places. All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Steve Totaro wrote: Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ When Digium starts using 1.4 in ABE then I would consider using it in a production environment. All I ever hear is soon, and I have heard that for months if not the whole year. Until Digium itself is comfortable selling and supporting this version, then neither am I. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] Why no port to Windos?
Windows is a half-baked, dying OS that in essence is a 32 bit extension and graphical shell, for a 16 bit patch to an 8 bit operating system, originally coded for a 4 bit microprocessor, written by a 2 bit company, that can't stand 1 bit of competition. Line of the year ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Saturday 15 December 2007 10:02:23 Rob Hillis wrote: One of the biggest barriers to upgrading are the number of little gotchas in syntax changes that can make an upgrade from 1.2 to 1.4 quite painful. After the pain I went through upgrading to 1.4, I've always been recommending to people to think twice about upgrading if 1.2 does what they require. Many of the changes may have been seen as minor - one or two changes are to be expected, but I ran into at least half a dozen - mostly variable changes if I recall correctly - things such as deprecating CALLERIDNUM in favour of CALLERID(num). Some of the breakage was minor (e.g. loss of caller ID processing) but some of them resulted in calls being dropped in unpredictable places. All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. -- Tilghman ___ --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 port to Windos?
Dovid B wrote: Windows is a half-baked, dying OS that in essence is a 32 bit extension and graphical shell, for a 16 bit patch to an 8 bit operating system, originally coded for a 4 bit microprocessor, written by a 2 bit company, that can't stand 1 bit of competition. Line of the year That joke (truth) is an old one actually. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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 port to Windos?
On Sat, 15 Dec 2007 08:30:09 +0100, randulo wrote: It's funny, but though I think nothing of having a linux box as a pbx, on 24/7 for years, I can't imagine using windows this way. I think there's little or no market for this whereas if there were a fanless, diskless embedded solution for just under $200 that came configured with the account (IAX or SIP and the proper provider) it would be a hit. For consumers, better to let them choose their own analog phone. For the teens, this adds their own line with unlimited dialing and international if needed. When appliances are down to this proce and they come pre-configured, plug it in, plug in a telephone and it works, that'll be the day this thing takes off. Even then, the market isn't huge. Maybe add in more intelligence in routing calls as an attraction. You nailed it Randy! When an Asterisk appliance and associated phones can compete with a Panasonic KXTG-4000 (or similar) on terms including price, ease of use reliabilitythat's when Asterisk for every grandma, aunt, uncle counsins (who never finished high school) will be viable for the broader home/residential market. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] Call Center Setup on asterisk
http://www.h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf - Original Message - From: satish patel To: asterisk-users@lists.digium.com Sent: Wednesday, December 12, 2007 2:09 PM Subject: [asterisk-users] Call Center Setup on asterisk Dear all I need call center setup on asterisk so i need do doucment and book .is it available on net PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org -- Looking for last minute shopping deals? Find them fast with Yahoo! Search. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Tilghman Lesher wrote: If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. A bit more prominent: yes. Every 10th usage: no. I wouldn't want gcc/perl/php/... to complain about deprecated syntax every 10th usage. IMHO that would be really confusing. And having to count those usages of deprecated things would mean additional overhead. Of course, all of these deprecations should be covered in UPGRADE.txt Definitely. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] DNS broken for www.voip-info.org ??
The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; DiG 9.4.1-P1 www.voip-info.org ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: SERVFAIL, id: 61402 ;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;www.voip-info.org. IN A ;; Query time: 4724 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sat Dec 15 11:54:57 2007 ;; MSG SIZE rcvd: 35 ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
I wonder if there are any major obstacles for upgrading. From our perspective I'd have to say package management. We manage a *lot* of asterisk boxes at client locations at the end of DSL connections. We have a schedule to make sure each box is updated once a month (e.g. these 10 boxes are updated in week 1 by Marcus, then in week 5 by Tom, etc.). If we can login and run a couple of simple commands to bring everything up to date, that saves us many hours every month. For better or worse, we generally use Gentoo Linux on our servers. With one command (emerge -DuavN world) I can bring a box completely up to date. Asterisk 1.2's portage packages are generally stable and fairly up-to-date. So, doing a portage update automatically upgrades asterisk, zaptel, libpri, speex and any other relevant packages at the same time as updating other core system libraries. Installing 1.4 is a pain. The individual installers for each relevant package have to be grabbed from Digium (or a mirror), then saved somewhere, then untarred, then ./configure'd, then made, then installed. And in a month's time if something's been updated, the procedure has to be repeated. It changes updating a server from a 5 minute operation into an hour or so. Yeah, part of it's laziness, but it's more about efficient use of employee time. If 1.2 does what the client needs and 1.4 would require many times the admin time, it isn't happening. In terms of fixing it - Digium could perhaps consider providing packages for the common *nix distros, which would be updated by them when new versions are released. We could then add the Digium layer (as it's referred under portage, other package managers probably call it something different) and it would be sync'd at the same time as the main distro portage tree. This is something I'd consider paying an annual subscription for. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ --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] DNS broken for www.voip-info.org ??
Steve Johnson wrote: The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? There are several mirrors, see http://www.google.com/search?q=cache:www.voip-info.org/wiki/index.php%3Fpage%3DVoip-Info%2BMirrors Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: On Saturday 15 December 2007 10:02:23 Rob Hillis wrote: One of the biggest barriers to upgrading are the number of little gotchas in syntax changes that can make an upgrade from 1.2 to 1.4 quite painful. After the pain I went through upgrading to 1.4, I've always been recommending to people to think twice about upgrading if 1.2 does what they require. Many of the changes may have been seen as minor - one or two changes are to be expected, but I ran into at least half a dozen - mostly variable changes if I recall correctly - things such as deprecating CALLERIDNUM in favour of CALLERID(num). Some of the breakage was minor (e.g. loss of caller ID processing) but some of them resulted in calls being dropped in unpredictable places. All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. Hello Tilghman, So if I read you correctly, all of the pain of the upgrade is due to lack of effort on the participants part! This seems a whole lot like the attitude of proprietary vendors when they don't want to support a feature that is outside the scope of what they want to maintain. I thought this was an open source project that would allow participants to have a voice in what is or isn't included in a new release. Even an non developing end user provides valuable benefit to the project in QA and bug information to improve the project as a whole. Most (With exceptions) projects have a bit more interest in what the user community wants or needs in a package. The attitude of this project seems to be If you want it code it yourself, however if it something that doesn't map to the ideas of what Digium wants then it will never make it into the official release. I don't understand why so much community support is placed into the project considering that the typical end user is treated like a second class citizen. So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. My 2 cents, ok 25 cents, Dave ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. We might want to be more informative about those changes. We need to make a clear list of things you need to start changing as a user of 1.4 to prepare for lost functionality in 1.6. This information already exist, but should maybe be a bit more public. In some cases we do have to change in a dramatic way and can't preserve the old functionality to solve a bug in the software. This requires thorough discussion in the developer group and is something we really want to avoid at all costs. If this happens, it's clearly documented in the software. Thank you for your feedback, it's important to us. /O ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Sat, Dec 15, 2007 at 06:11:47PM -, Chris Bagnall wrote: I wonder if there are any major obstacles for upgrading. From our perspective I'd have to say package management. We manage a *lot* of asterisk boxes at client locations at the end of DSL connections. We have a schedule to make sure each box is updated once a month (e.g. these 10 boxes are updated in week 1 by Marcus, then in week 5 by Tom, etc.). If we can login and run a couple of simple commands to bring everything up to date, that saves us many hours every month. For better or worse, we generally use Gentoo Linux on our servers. With one command (emerge -DuavN world) I can bring a box completely up to date. Asterisk 1.2's portage packages are generally stable and fairly up-to-date. So, doing a portage update automatically upgrades asterisk, zaptel, libpri, speex and any other relevant packages at the same time as updating other core system libraries. Actually, it isn't that up-to-date. Installing 1.4 is a pain. The individual installers for each relevant package have to be grabbed from Digium (or a mirror), then saved somewhere, then untarred, then ./configure'd, then made, then installed. And in a month's time if something's been updated, the procedure has to be repeated. It changes updating a server from a 5 minute operation into an hour or so. Yeah, part of it's laziness, but it's more about efficient use of employee time. If 1.2 does what the client needs and 1.4 would require many times the admin time, it isn't happening. How about you coming up with an 1.4 package? I'm willing to help anybody with zaptel 1.4 and a bit with Asterisk. Packages should be integrated with the distribution, or else they break the upgrade. If there are enough users, one of them should be able to do that. If there aren't enough users, well, I'm not sure who will bother... -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
At 10:14 AM 12/15/2007, you wrote: So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. My 2 cents, ok 25 cents, Dave I have a somewhat different opinion. I once used a product where upon announcing the features of the next release to a rather disenchanted audience pointed out that the product now contained what you needed to get your job done, and not necessarily what you wanted. The people unwilling to learn walked away disappointed, those willing to think about what they'd been given were constantly surprised at the newfound capabilities of the product. People always ask for stuff they don't need that they're unwilling to figure out how to do themselves. A perfect example is the new dial plan function array(), it has nothing to do with arrays, doesn't accomplish anything useful that couldn't have been done by allowing commas in set(), or calling it setmany(), and means if real arrays ever get added to the language we have to come up with new function names while the obvious one has already been taken to mean something not related. I know lots of people have thousands of hours in dial plans to solve specific problems, but personally I'd have no problem if the deprecated the whole dial plan script language and started over. Well, I'd be happy if they came up with an elegant language with functions, parameters and proper variable scoping while getting rid of line numbers and all the rest of the baggage that shouldn't have been there in the first place. AEL is an attempt to solve some of that, but as it's just a precompiler to the underlying language it has limitations that shouldn't be there. I'm sure that's not a popular opinion as people don't tend to like change, but in the long run it would make Asterisk a better product. Sadly it's probably already too late. I still cringe whenever I see people acknowledge line numbers in new books. Ira ___ --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] DNS broken for www.voip-info.org ??
DNS for www.Voip-info.org should be back online shortly. In the meantime here are mirrors: a.. SimpleVoip.info - Location: California, USA, Bandwidth: 100M, Updated Nightly b.. Malico Inc. - Location: Tao-Yuang, Taiwan, Bandwidth: 1Mbps, Updated Daily c.. Totalip - Location: Oslo, Norway. Bandwidth: 100Mbit, Updated Daily d.. http://www.telephreak.org/voip-info - Location: Florida, USA. Bandwidth: Dual DS1, Updated hourly. e.. AFOYI Mirror - Location: Adelaide, Australia, Bandwidth: 12Mbps, Updated 3 hourly f.. LinuxSystems - Location: Florida, Bandwidth: 10M, Updated Nightly Thanks for using voip-info.org! [EMAIL PROTECTED] - Original Message - From: Steve Johnson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 15, 2007 7:57 AM Subject: [asterisk-users] DNS broken for www.voip-info.org ?? The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; DiG 9.4.1-P1 www.voip-info.org ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: SERVFAIL, id: 61402 ;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;www.voip-info.org. IN A ;; Query time: 4724 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sat Dec 15 11:54:57 2007 ;; MSG SIZE rcvd: 35 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Hello All , On Sat, 15 Dec 2007, Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ The one item mentioned in some of the responses to the thread that this message started is the modification of commands (dialplan others) , variables and such . Tilghman mentioned these changes are collected in UPGRADE.txt . But (I have to admit IMO) , The procedure necessary to follow to get a system running 1.4 is not a upgrade path . It is a migration . ie: duplicate the system(s) running 1.2 successfully today onto seperate hardware make the changes necessary to create a (near as possible) functioning system as the present systems then swap them . If this path was more of a true upgrade path then 1.4 would probably be used far more than 1.2 . Hth , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | [EMAIL PROTECTED] | Fairbanks, AK. 99701 | only on AXP | +--+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 compilation
Dear Kiven; Actually it is default and not degault. Also, I was doing the compilation remotely via the Putty. Another thing, I did another senario and got another thing, as below: I copied /usr/local/lib to /usr/lib and then I restarted asterisk, but when I come back to run it, then it was giving error that Segmentation error or Segmentation fail, actually I did not remeber it exactly, but was something related to segmentation. Then, I moved to the site where Asterisk existed and I decided to recompile h323 and then asterisk, when I run the make and make opt at the server it self and under the directory: /usr/src/asterisk-1.4/channels/h323, it was take the commands without error but does not give any text output (messages), I do not know if that good indication or not, then I compiled asterisk again, and it worked fine. Till now, I do not know why it was giving me default and does not know how to know if chan_h323 is really working or not, how can I test? Is it by establishing h323 trunk? Regards Bilal cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' This is *exactly* what showed up on your session? The word 'degault' does not appear in the Makefile at all, so if that is the message that you got then your source tree is corrupted. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --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] Upgrade to Asterisk 1.4 - from a 1.0 style configuration
It seems that all the warnings about deprecated functions in 1.2 did not give the desired effect - that users move from the 1.0 commands to the new applications and functions in 1.2. That caused real problems when going from 1.2 to 1.4, since the dialplans where still on 1.0 level, not 1.2 level. I feel that it seems very important to help people go from a 1.0 configuration running in a 1.2 system to a 1.4 system prepared for 1.6, so that the move to 1.6 will be an upgrade, not a complicated migration. Going directly from 1.0 to 1.4 style configuration without using any of the deprecated commands in 1.4 will make upgrade to 1.6 much easier. While the rest of the discussion is interesting (new dial plan language etc) it's a bit off topic (as I indicated in the mail that started this thread). Thank you for all the feedback. There's a lesson to be learned here. Since the 1.0 configuration still works in 1.2, users has not upgraded their configurations to the new platform but use it in backwards compatibility mode. The path of least resistance :-) I'm still eager for more feedback, so keep the mails coming - either to the list or directly to me. /O ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
When Digium starts using 1.4 in ABE then I would consider using it in a production environment. All I ever hear is soon, and I have heard that for months if not the whole year. Until Digium itself is comfortable selling and supporting this version, then neither am I. Steve, That's very good feedback. Let's try to find out what's holding them. /O I can tell you that Digium's response is going to be real soon... I just started using 1.4.X. Wish me luck ;) ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. I don't think there is an issue with people reading UPGRADE.txt as there is re-doing all of our custom code. ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Tilghman Lesher wrote: If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. After thinking about it for a while this is not true. Well, it's true for the dialplan. Changing CALLERIDNUM to CALLERID(num) is easy. But i guess people use a lot of custom applications built around Asterisk 1.2. If any of the interfaces (AGI, AMI, CDRs, queue log, ...) change that might break the app. Fixing these apps might not be trivial and probably requires a lot of fine-tuning. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. Olle, I can explain why *we* haven't upgraded our customers yet: fear and lack of necessity. 1. Fear of new bugs (as we encountered during the 1.2 lifecycle). It will be a *lot* of work to test any and all feature our customers use before we install it on their premises. 2. Fear that we will find out the hard way that the UPGRADE.txt file maybe was correct and complete when 1.4.0 came out, but is now lagging behind. 3. Fear that we will find out the hard way why Digium is not supporting it yet commercially. 4. I have been following all developer fixes since 1.4.0. Lately I have been too busy to read them, but there were - to my taste - far too many fixes for crashes or otherwise important bugs. I'll wait until the stream of fixes per month slows down to a trickle. 5. Lack of necessity. Not enough features that will convince our customers to pay for the upgrade. Other people have mentioned the upgrade path. I can imagine that is an important point as well, but not for us, as our customers can upgrade their system with a yum upgrade. This saves our support people a lot of time, and thus the customers a lot of money. How can the uptake for 1.4 be improved? No it's not the bugs. I'd suggest some marketing. Create a section on the website on the upgrade from 1.2 to 1.4 with: - short list of most important features - an extensive guaranteed-to-be-complete categorized list of all changes. - some explanations on stability (I mean the amount and type of fixes that went into the releases since 1.4.0, and how it is dying down now) And of course Digium could create a lot of trust by commercially supporting 1.4. Thanks, Ron Arts NeoNova - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 smime.p7s Description: S/MIME Cryptographic Signature ___ --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 port to Windos?
For this market, people don't want anything complicated. I would imagine the software equivalent of a run-of-the-mill answering machine. Which has existed, in one form or another, for years. I was using a voice enabled faxmodem a decade ago to answer my phone. The software that came with it (don't remember the name, but WinFax also does/did this) even allowed for a simple IVR, for mailbox selection and whatnot. The only things it didn't do that asterisk does (and would be useful to the average Joe) was support multiple phones/extensions and send voicemail messages via email. I just don't see the consumer market opening up enough to make it worth the expense and hassle of writing/supporting Windows drivers. Digium is great at what they do - I wouldn't want them changing what they're doing now if it could impact the quality of their core products/drivers. St- ___ --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 port to Windos?
Which has existed, in one form or another, for years. I was using a voice enabled faxmodem a decade ago to answer my phone. The software that came with it (don't remember the name, but WinFax also does/did this) even allowed for a simple IVR, for mailbox selection and whatnot. The only things it didn't do that asterisk does (and would be useful to the average Joe) was support multiple phones/extensions and send voicemail messages via email. I think what you are looking for is named SuperVoice : http://www.supervoice.com/asp/products_supervoice_fax_products.asp I was using it to receive faxes and voicemail. It didn't email me my fax and/or voicemail but it would page me the number of faxes and voicemail I had everytime it received one or the other. The only problem I was having was that, since running on Win9x, sometime my phone line would stay busy and that would signal the time to go home and reboot the computer. Maybe some people would like Asterisk for windows, but I would not touch it with a ten foot pole :) ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
- Original Message - From: Ira [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 15, 2007 2:50 PM Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! At 10:14 AM 12/15/2007, you wrote: So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. My 2 cents, ok 25 cents, Dave I have a somewhat different opinion. I once used a product where upon announcing the features of the next release to a rather disenchanted audience pointed out that the product now contained what you needed to get your job done, and not necessarily what you wanted. The people unwilling to learn walked away disappointed, those willing to think about what they'd been given were constantly surprised at the newfound capabilities of the product. People always ask for stuff they don't need that they're unwilling to figure out how to do themselves. A perfect example is the new dial plan function array(), it has nothing to do with arrays, doesn't accomplish anything useful that couldn't have been done by allowing commas in set(), or calling it setmany(), and means if real arrays ever get added to the language we have to come up with new function names while the obvious one has already been taken to mean something not related. I know lots of people have thousands of hours in dial plans to solve specific problems, but personally I'd have no problem if the deprecated the whole dial plan script language and started over. Well, I'd be happy if they came up with an elegant language with functions, parameters and proper variable scoping while getting rid of line numbers and all the rest of the baggage that shouldn't have been there in the first place. AEL is an attempt to solve some of that, but as it's just a precompiler to the underlying language it has limitations that shouldn't be there. I'm sure that's not a popular opinion as people don't tend to like change, but in the long run it would make Asterisk a better product. Sadly it's probably already too late. I still cringe whenever I see people acknowledge line numbers in new books. Ira Then fork or learn how to re-write and submit your code for this elegant language you speak of and see how it flies. Thanks, Steve Totaro ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Saturday 15 December 2007 13:50:35 Ira wrote: A perfect example is the new dial plan function array(), it has nothing to do with arrays, doesn't accomplish anything useful that couldn't have been done by allowing commas in set(), or calling it setmany(), and means if real arrays ever get added to the language we have to come up with new function names while the obvious one has already been taken to mean something not related. I'm curious to hear how you would have approached the problem of retrieving multiple columns out of a database and setting each column to its own variable. That is precisely what ARRAY() is designed to accomplish, and it CANNOT be done by letting Set have multiple key/value pairs. -- Tilghman ___ --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] One touch with Polycom Phones.
Hi Everyone,
I am attempting to migrate my org to Polycom desktop phones. I need to
find a one touch park method.
I am using SIP 2.2.0.0047, and BootRom 4.0.0.0.423
I have found a few methods in previous posts:
Like this one posted by Anthony Rodgers
http://groups.google.com/group/Asterisk-users/browse_thread/thread/e86bbed2926294df/76a7d6165b37f04c?lnk=stq=asterisk+polycom+park#76a7d6165b37f04c
[internal] ; or whatever the relevant context is for you - it's usually
wherever your *Polycom* lives
include = parkedcalls
exten =
callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
The above extension seems to work, but my Polycom asks via the LCD for
an input before performing the action, removing the One Touch aspect.
Does anyone have any ideas how to setup a park button on Polycom that
works with Asterisk?
Thanks everyone,
Matt
begin:vcard
fn:Matthew Mackes
n:Mackes;Matthew
org:Delta Sonic Car Wash Systems;IT Department
adr;dom:;;570 Delaware Ave;Buffalo;NY;14202
email;internet:[EMAIL PROTECTED]
title:Network Administrator
x-mozilla-html:TRUE
url:http://www.deltasoniccarwash.com
version:2.1
end:vcard
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[asterisk-users] Reputable company for SIP/IAX2 trunking
Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have it dial individual peers according to an extension. Looking for some feedback on what provider is quality these days. I don't mind paying an extra dollar or two. Thanks, - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
gateway.
The issue is that my cellphone shows my PBX's number, not the original
calling number.
My dialplan looks like:
[globals]
...
TRUNK=SIP/sip_proxy-out
CELL=${TRUNK}/208xxx
PHILIP=SIP/bedroom_1SIP/office_2SIP/kitchen_1${CELL}
[incoming]
exten = s,1,Answer()
; sometimes signaling and media get out of sync on cell networks...
exten = s,n,Wait(0.75)
exten = s,n,Playback(main-menu)
exten = s,n(exten),Background(vm-enter-number-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup
...
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
exten = 112,1,Macro(stdexten,112,${REDFISH})
exten = 112,n,Goto(s,exten)
...
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
Ok, so far, so good.
The problem is that when we hit Macro(stdexten,111,${PHILIP}) and it
does the Dial(${PHILIP}) which includes the
SIP/sip_proxy-out/208xxx, 208xxx rings with my PBX's extension.
Oddly, the internal phones ring with outside caller's extension.
[sip_proxy-out]
type=peer
fromuser=208nnn
fromdomain=x.x.x.x
host=y.y.y.y
call-limit=5
nat=yes
So I'm not setting the callerid on the peer by default. What am I
missing? Do I need to modify the stdexten macro to dial with the 'o'
option? Or can I set this explicitly with a 'Set' before calling the
macro? Or do I need to be missing with the RDNIS?
Oh, I'm running Asterisk 1.2.25... (yes, I'll upgrade when AstLinux
upgrades).
-Philip
P.S. I tried adding |o to the end of the PHILIP variable, but this
didn't seem to make a difference.
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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Olle E Johansson wrote: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. We might want to be more informative about those changes. We need to make a clear list of things you need to start changing as a user of 1.4 to prepare for lost functionality in 1.6. This information already exist, but should maybe be a bit more public. In some cases we do have to change in a dramatic way and can't preserve the old functionality to solve a bug in the software. This requires thorough discussion in the developer group and is something we really want to avoid at all costs. If this happens, it's clearly documented in the software. Thank you for your feedback, it's important to us. /O Along that this same line, I ran 1.0.something for a long time and it was working just fine for my SOHO. I had a channel bank to interface pots lines from the local Telco and feed the analog phones in the house. Over time, I replaced most of those analog phones with SIP phones. An unfortunate incident caused us to lose that server and several sip phones. When I recovered enough to rebuild *, I tried 1.4 and it would not compile completely and zaptel did not load properly. I download 1.2 and it worked with the same configs as 1.0, but the quality was poor. That was due to hardware issues. I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 with the then current libpri and zaptel and the call quality came back. But I had a hard time with syntax changes. Basically I was jumping from 1.0.x to 1.4.x in one leap. My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. Lyle As it is syntax problems or changes are not noticed or logged until Asterisk tries to execute them. If there is a chunk of code that is only hit once a week??? It almost came to a point of scraping Asterisk because of the push back from the family. ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
You've hit the nail on the head with the crux of the pain I went through. Finding stuff that was broke that I didn't realise was broke until someone bothered to tell me about it. I'm sure everyone is familiar with just how often users report problems caused by themselves, but don't report stuff caused by a problem that needs to be fixed. Not being a regular user of the PBX, I didn't realise the problem existed until a few weeks later when someone whined about something being broken for ages, despite my repeated pleas for people to report problems to me. Lyle Giese wrote: My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. ___ --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] Trixbox Arbitrary Command Execution Vulnerability
A set of scripts were recently discovered in the trixbox line of PBX products, which connect to a remote host every 24 hours, to retrieve an arbitrary list of commands to be executed locally. These scripts were added under the guise of submitting 'anonymous usage statistics', however, with the help of DNS pollution, or malice on the part of the sponsoring company (Fonality), all up-to-date versions of trixbox could be instantly disabled, or worse. According to trixbox Community Director, Kerry Gerrison, a new version of trixbox will be available by December 18th which will allow you to 'opt-out' (meaning that it will still be enabled by default) of this behavior. Further details: http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home http://www.trixbox.org/trixboxs-new-hardware-audting-tool _ Share life as it happens with the new Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_122007___ --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 B2BUA and Site to Site transfers
Hi, Thanks very much for your response. I'm don't think setting reinvites on will fix your problem. The only thing I can think of is that you use some sort of call parking to park the call on SiteB's asterisk server and then have the person at siteB pick up the call from the parking lot I didn't consider call parking... I'll see if that is suitable for the users.. Anyone else know a better way to do this? Would be good to hear any other ideas. Thanks, Chris Bennett (cgb) ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. I fully agree that a preprocessor that checks the configuration would be wonderful, much like Apache's configtest. Let's hope a developer that knows the internal parsing code can jump on that idea and make it happen. Right now you can run asterisk -cv | tee /tmp/debug with Asterisk 1.4 to get all the messages about errors in all configuration files except the dialplan and files that are loaded when needed (like manager.conf and meetme.conf). asterisk -c starts Asterisk in the foreground and outputs all messages to the console, things that you may not catch otherwise when you start Asterisk in the background. Thanks for your suggestion, I'll make sure it's brought forward. Can't promise that it will happen, since this is an open source project. We rely on the developers and the ones that fund the developers. Regards, /Olle ___ --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] LDAPget question, usage
Hi,
I've recently come across LDAPget (version 2.0rc1) and I've been trying to
get it functional in my test environment (Asterisk 1.4.15 and MS Active
Directory 2003) but I can't seem to get it working.
I put together a test extension to try to change the CALLERID(name) by way
of a LDAP query to AD:
extensions.conf
exten = 100,1,Answer()
exten = 100,n,LDAPget(CALLERID(name)=cidname)
exten = 100,n,SayPhonetic(${CALLERID(name)})
exten = 100,n,Hangup()
ldap.conf
[cidname]
host = dc1.test.domain.lan
version = 3
user = cn=asterisk,ou=services,dc=test,dc=domain,dc=lan
pass = xxx
base = ou=Addressbook,o=test.domain.lan
filter = ((objectClass=person)(|(telephoneNumber=${CALLERID(num)})))
Whenever this is tested, the following is output on the Asterisk console:
[Dec 16 02:24:04] WARNING[5258]: app_ldap.c:266 ldap_lookup: LDAPget: search
failed: Operations error
I'm the first to admit I'm not an LDAP or a MS guy, so if my query is wrong
I'll happily slink away. It does run in AD at least, and I'm certain
authentication is working correctly. I've checked to make sure both my
CALLERID(num) and LDAP telephoneNumber match exactly as well.
Is anyone doing something similar? Am I missing something?
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