[asterisk-users] PCI32 and PCI-X compatibility
Hi, this is my 1st message, I'm writing to ask if anyone knows if a PCI32 card like the TDM400P (quad analog) or the B410P (quad BRI) is working on a PCI-X bus, at 100MHz or 133 MHz. I'm really stuck with this, since I found a partial yes on this mailing list but my supplier says no! Thanks, Marco ___ -- 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] Friday Feb 15th @ 12 Noon EST: VoIP Users Conference welcomes Lumenvox
This Friday, February 15th, at 12 Noon EST, 9AM PST, 17:00 UTC, Lumenvox will be joining us on the VoIP Users Conference. This week, the last in a series about IVR, Lumenvox will be there to discuss and field your questions on their speech recognition solutions. http://www.VoipUsersConference.org - for info on the conference, how to connect, etc IRC freenode.net #voip-users-conference - to ask questions and chat if you do not wish to talk http://food4wine.ning.com - VoIP Users Conference Community site (blogs, forum, notes, archives) In a nutshell, you can just call in via PSTN beginning at Noon EST: Phone Number: (724) 444-7444 Upon answer, enter 22622# 1# or see the voipusersconference.org for SIP and Talkshoe details. You can also see all the records here: http://www.talkshoe.com/talkshoe/web/talkCast.jsp?masterId=22622 Talkshoe has a chat/SIP client combo you can download for WIndoze/Mac if that's of interest to you. It makes following the discussion easier. Here is what that looks like: http://tinyurl.com/3c6ztn ___ -- 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] restart asterisk daily
On Feb 13, 2008 8:48 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Actually, I donno it is a memory leak or not. I have a server only running asterisk. As time goes by, the free memory shown in the top is decreased. After I restart the asterisk, the free memory comes I observed the same behavior. Someone told me that that's a normal feature of linux, it manages memory that way. If that's true, than it isn't normal to see the same (large) amount of free memory over time on a box running asterisk only. However, I rarely restart and it hasn't caused problems. Here's mine right now: 09:14:39 up 73 days, 18:47, 2 users, load average: 0.00, 0.00, 0.00 67 processes: 65 sleeping, 2 running, 0 zombie, 0 stopped CPU states: 0.1% user 3.9% system 0.0% nice 0.0% iowait 95.8% idle Mem: 515460k av, 509416k used,6044k free, 0k shrd, 80052k buff 152896k active, 191472k inactive Swap: 477248k av, 0k used, 477248k free 242404k cached 'course, these days, half a meg isn't much :) ___ -- 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] restart asterisk daily
On Wed, Feb 13, 2008 at 03:48:14PM +0800, Rilawich Ango wrote: Actually, I donno it is a memory leak or not. I have a server only running asterisk. As time goes by, the free memory shown in the top is decreased. After I restart the asterisk, the free memory comes again. That's why I wonder if regular restart asterisk is necessary. Use a crontab to restart asterisk is a way to do it but you have to maintain a crontab. Is it possible to use logrotate instead? Or other better way? [EMAIL PROTECTED]:~$ free -m total used free sharedbuffers cached Mem: 485477 7 0 0100 -/+ buffers/cache:376108 Swap: 1419270 1149 [EMAIL PROTECTED]:~$ top -b | head -n 5 top - 10:18:32 up 19 days, 14:38, 24 users, load average: 0.08, 0.33, 0.21 Tasks: 166 total, 1 running, 163 sleeping, 2 stopped, 0 zombie Cpu(s): 1.1%us, 0.1%sy, 0.0%ni, 98.2%id, 0.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem:496648k total, 489044k used, 7604k free, 32k buffers Swap: 1453840k total, 276740k used, 1177100k free, 103380k cached [EMAIL PROTECTED]:~$ ps aux | grep asterisk asterisk 9559 0.0 2.5 474896 12892 ?Ssl Feb12 0:00 /usr/sbin/asterisk -p -U asterisk Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! In fact: 1. The system has some 100MB of free memory. almost all of it is used for caching and such. 2. Asterisk overcommits memory: it generally asks the kernel huge ammounts of memory, but doesn't really try to use them. At least with Linux such overcommits are not claimed at all. -- 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
[asterisk-users] differences
Hi All What are the differences between asterisk 1.2.4 and 1.4.6 beta In functionality ,services and bugs. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] How to soft hangup all channels at a time .
On Wed, Feb 13, 2008 at 01:49:38PM +1100, Mohammad Salaque wrote: Dear all, Anyone can point me how to soft hangup all channels using single command ? I am using Asterisk 1.4.15. restart now -- 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] differences
On Mon, Feb 11, 2008 at 05:25:44PM +0200, Khaled Chehab wrote: What are the differences between asterisk 1.2.4 and 1.4.6 beta You probably ask about Asterisk 1.4 vs. Asterisk 1.6 beta, right? In functionality ,services You can probably read about some of the changes in the file UPGRADE.txt . http://svn.digium.com/svn/asterisk/trunk/UPGRADE.txt and bugs. Bugs? You mean undocumented features? ;-) Some of them are known to be documented in http://bugs.digium.com/ -- 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] Realtime SIP peers - reloading cached info
That's why I didn't see anything about the REALTIME function when I went looking - many of our production systems are still on later versions of 1.2. Given that it wasn't made obsolete at the /beginning/ of the 1.4 cycle, I'm hoping Digium reconsider making it obsolete in 1.6 and schedule it for removal in 1.8. Half a development cycle isn't a very long time for a warning that a function will be removed. Atis Lezdins wrote: On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If it is being removed in 1.6, I'm a little concerned since there's no mention of this when you show the application, nor on voip-info.org. What application/function is it being replaced by? There's an obsolete warning in 1.4.18, but i somehow remember that it's obsolete already since some 1.4.11 It's func_realtime as i said before. usage shouldn't be much different, you can replace with: Set(REALTIME(sip_buddies,name,100,my_field)=foo); Also, seems that func_realtime will soon support SQL INSERT's and DELETE's :) Regards, Atis Atis Lezdins wrote: | On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote: | -BEGIN PGP SIGNED MESSAGE- | Hash: SHA1 | | Atis Lezdins wrote: | | By RealTimeUpdate do you mean func_realtime? It shouldn't care, as | | cache is not implemented in realtime level, but higher (chan_sip). | | | | Are you sure you need sip show XXX load. If you sip prune peer | | data, it should be re-loaded on next access. | | | | What i was suggesting - to dig into chan_sip and create dialplan | | application SipPrune(peer) that would prune the peer directly, by | | using corresponding function - sip_prune_peer() in chan_sip.c - that | | way you will gain some extra performance, as there's no manager/cli | | overhead. | | | | However if you're uncomfortable with C, the app_system shouldn't cause | | any troubles :) | | RealTimeUpdate is more likely to correspond to app_realtime rather than | func_realtime. | | As to my knowledge - that is obsolete and being removed in 1.6, | func_realtime replaces it. That's why i wondered about name - I just | never happened to use it :) | | Regards, | Atis | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Remi - http://enigmail.mozdev.org iD8DBQFHsnaM6uKn5cBSgGQRAo/TAKDCruPrn2nm2XV/PYbfSuBKA0j5OwCfQ/Ox QE3SYEmZ01QHUT4ITwmLnT0= =SKEW -END PGP 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 ___ -- 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] urgent-channels
Hi All I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No 215.96.142.83(None) caac0846-cf 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/0 unkn No 85.219.172.253 (None) 67a0d6b3191 00101/0 unkn No 85.219.172.253 (None) 0d778c314f5 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 30a7d77c5bc 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 569 active SIP channels Why these channels exit or didn't be killed,how can I solve that. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] HP proliant and hpasm
On Feb 10, 2008 2:01 AM, Steven [EMAIL PROTECTED] wrote: Is anyone successfully running asterisk on an HP proliant while using their management software, hpasm? I have two DL360's and two TE220B's. The cards have their own IRQ's. No matter what combination of settings I use, the cards fail the patlooptest if hpasm (ver 7.9.1) is running. If I stop it the cards pass the test. Hi there, we do run the hpasm on the HP Proliant servers without any problem. We had some issues a while ago with ML350's that kept giving problems, IRQ misses, red alarms, dropped calls, etc.. Everything 'looked' fine (no irq sharing etc..) but the problem was related to iLO. Disabling iLO made it all work.. So if you have issues, try that for starters.. What distro and versions are you using? cheers, stoffell ___ -- 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] Hardware needed
Dear List, I have to plan an instalation of an asterisk box for over 400 extensions (Sip and Iax2) and 4 PRI channels. I do not know which hardware (server) should I buy to support this amount of extensions. Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- 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] Hardware needed
On Feb 13, 2008 10:15 AM, voip crazy [EMAIL PROTECTED] wrote: Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? You will probably get some useful info on the list but also check out voip-info.org: http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations cheers, stoffell ___ -- 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-dev] chan_ooh323 patches compatible with codec negotiation patch applied to asterisk 1.4.17
Hi all, It is posted here: http://bugs.digium.com/view.php?id=11976 Still waiting for the approval. Please see the notes. thanks, Ganbold On 2/12/08, Johan Wilfer [EMAIL PROTECTED] wrote: Ganbold Tsagaankhuu wrote: Hi all, Sorry for cross posting. I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec negotiation patch changes applied to asterisk-1.4.17. Please let me know whether my patches are correct or not. thanks in advance, Ganbold For licensing issues nobody will be able to use your patch if you don't submit it thought the bug tracker at http://bugs.digium.com/ You will be able to agree to the digium license after you have created an account. There is also a bug tracker introduction that is useful to read at http://asterisk.org/developers/bug-guidelines Nice work! /Johan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ -- 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] urgent-channels
I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No 215.96.142.83(None) caac0846-cf 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/0 unkn No 85.219.172.253 (None) 67a0d6b3191 00101/0 unkn No 85.219.172.253 (None) 0d778c314f5 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 30a7d77c5bc 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 569 active SIP channels asterisk1*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls Why these channels exit or didn't be killed,how can I solve that. Regards _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DTMF dialing
On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. I have just double check and my phones use DTMF in RFC2833 mode. I wil try to downgrade my zaptel later today -- Andres Jimenez GPG : http://www.andresin.com/gpg/[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] Automatically start after restart
Dear Matt; Special thanks for you, but I did not understand what u mean by: Hash: SHA1? Do u mean to type SHA1 from the putty when I am connected remotely? I tried that and I did not find such command, but rather I found commands like sha1sum, sha224sum, sha256sum, ... Can u advise what exactly meant by SHA1 and from where to be typed? I am using Fedora core 7. Regards Bilal -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 bilal ghayyad wrote: Hi All; How can I let Asterisk start automatically once the machine restarted without need to type asterisk -cvvv? Any script or something that can do that? Also, in which command line screen (F1 or F2 or F3 or ..?) I will find it? Use the asterisk init scripts or safe_asterisk: 1. type make config after you finish compiling and installing Asterisk 2. type service asterisk start (in Fedora/CentOS etc) or /etc/init.d/asterisk start in other distros 3. type asterisk -r to connect to the process or do the same but using safe_asterisk instead of the scripts. The benefit of the make config stuff is that you can then do chkconfig asterisk on to make Asterisk start up automatically on boot. - -- Kind Regards, Matt Riddell Director 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] Automatically start after restart
On 2/13/08, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Matt; Special thanks for you, but I did not understand what u mean by: Hash: SHA1? Do u mean to type SHA1 from the putty when I am connected remotely? I tried that and I did not find such command, but rather I found commands like sha1sum, sha224sum, sha256sum, ... Can u advise what exactly meant by SHA1 and from where to be typed? ROFL Sorry, can't stop laughing... Bilal, you should first learn netiquette, and read email completely. You would find then, that answer is enclosed inline (as it's commonly done in emails). SHA1 indicates hashing algorithm for signed email. Regards, Atis I am using Fedora core 7. Regards Bilal -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 bilal ghayyad wrote: Hi All; How can I let Asterisk start automatically once the machine restarted without need to type asterisk -cvvv? Any script or something that can do that? Also, in which command line screen (F1 or F2 or F3 or ..?) I will find it? Use the asterisk init scripts or safe_asterisk: 1. type make config after you finish compiling and installing Asterisk 2. type service asterisk start (in Fedora/CentOS etc) or /etc/init.d/asterisk start in other distros 3. type asterisk -r to connect to the process or do the same but using safe_asterisk instead of the scripts. The benefit of the make config stuff is that you can then do chkconfig asterisk on to make Asterisk start up automatically on boot. - -- Kind Regards, Matt Riddell Director 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] Realtime SIP peers - reloading cached info
13 feb 2008 kl. 10.27 skrev Rob Hillis: That's why I didn't see anything about the REALTIME function when I went looking - many of our production systems are still on later versions of 1.2. Given that it wasn't made obsolete at the beginning of the 1.4 cycle, I'm hoping Digium reconsider making it obsolete in 1.6 and schedule it for removal in 1.8. Half a development cycle isn't a very long time for a warning that a function will be removed. First, it's not Digium - it's the Asterisk developer team. There still is a difference, not all of us are employed by Digium. My work is mostly funded by myself nowadays, and some by customers that hires me as a consultant for various Asterisk projects. I tried to get more general funding to spend more time with Asterisk development, but failed. So please rememner that there are a few independent regular Asterisk developers out there that is not on the Digium payroll and still take part in decisions about Asterisk. The way it works is that we decide which functions to deprecate during the development cycle. So any decisions was made before the 1.4 release and stays for the duration of the 1.4 release. We did not deprecate anything in 1.4 after the initial release late 2006. The functionality that was marked as deprecated in 1.4 will be removed in 1.6. In fact, it's propably already removed in the development code that is the base for the future 1.6. Over a year is a long time for a warning like this, considering that 1.6 won't be out for a while (we're in beta test cycle) it might even be 1.5 year warning. That should be more than enough for most people - I hope. Considering that people don't upgrade quickly, it will propably be more than that for most users (as you are still on 1.2 :-) ) Just wanted to clarify the process, I have no detailed insight into the realtime functions. /O --- * 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] How to detect if SIP extension BUSY?
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote: My problem is in subject. As I read in documentations and voip-info.org I can't user ChanIsAvalil because it not detects BUSY information on SIP channel. I've tried to use SIPPEER function, but it gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP extensions are Linksys PAP2. Have any other idea? Well? Is it impossible to detect BUSY on SIP channels? -- Best regards, Gergomailto:[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
[asterisk-users] Attendant phone
Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? Thanks, VoipCrazy ___ -- 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] Attendant phone
On 2/13/08, voip crazy [EMAIL PROTECTED] wrote: Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? You want 200 LEDs on single phone? Wouldn't it be wiser to have some web app that shows you those states by groups, etc.. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] Attendant phone
The norm (if memory serves) is about 64 70 extensions per attendant. After that, people usually split off onto multiple attendants just so the receptionists dont kill themselves in queues. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of voip crazy Sent: 13 February 2008 02:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Attendant phone Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? Thanks, VoipCrazy ___ -- 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] how to create a standalone voicemail server
On Mon, 11 Feb 2008 00:24:14 +, Cheikhou DIAW [EMAIL PROTECTED] wrote: i've been googling all night looking for a tutorial that shows how to make an asterisk standalone voicemail server , no way ! Asterisk: The Future of Telephony, Second Edition http://downloads.oreilly.com/books/9780596510480.pdf ___ -- 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] urgent-channels
A quick look at http://ftp.digium.com/pub/asterisk/releases/ tells me that 1.2.4 *might not* be the latest release of software. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: 13 February 2008 09:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] urgent-channels I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No 215.96.142.83(None) caac0846-cf 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/0 unkn No 85.219.172.253 (None) 67a0d6b3191 00101/0 unkn No 85.219.172.253 (None) 0d778c314f5 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 192.168.8.106(None) 30a7d77c5bc 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) efa10246-ea 00101/0 unkn No 192.168.8.106(None) 94910146-46 00101/0 unkn No 569 active SIP channels asterisk1*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls Why these channels exit or didn't be killed,how can I solve that. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned: It waits until the
script ends before moving on to the next step, which is Dial():
===
exten = s,1,AGI(netcid.py|${CALLERID(num)}|${CALLERID(name)}) exten
= s,n,Dial(${MYPHONE},5)
===
# cat netcid.py
#!/usr/bin/python
import socket,sys,time,os
def sendstuff(data):
s.sendto(data,(ipaddr,portnum))
return
sys.stdout = open(os.devnull, 'w')
if os.fork():
#BAD? sys.exit(0)
os._exit(0)
else:
now = time.localtime(time.time())
dateandtime = time.strftime(NaVm/%y NaVM, now)
myarray = []
myarray.append(STAT Rings: 1)
myarray.append(RING)
myarray.append(NAME + cidname)
myarray.append(TTSN Call from + cidname)
myarray.append(NMBR + cidnum)
myarray.append(TYPE K)
s = socket.socket(socket.AF_INET,socket.SOCK_DGRAM)
s.setsockopt(socket.SOL_SOCKET,socket.SO_BROADCAST,True)
portnum = 42685
ipaddr = 192.168.0.255
for i in myarray:
sendstuff(i)
#Must pause, and send IDLE for dialog box to close
time.sleep(5)
sendstuff(IDLE + dateandtime)
===
In another forum, people told me that I should fork twice. Is that
really necessary?
http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/278731
Thank you.
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Re: [asterisk-users] How to detect if SIP extension BUSY?
13 feb 2008 kl. 13.14 skrev Gergo Csibra: Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote: My problem is in subject. As I read in documentations and voip-info.org I can't user ChanIsAvalil because it not detects BUSY information on SIP channel. I've tried to use SIPPEER function, but it gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP extensions are Linksys PAP2. Have any other idea? Well? Is it impossible to detect BUSY on SIP channels? Place a call to it and if the phone reports BUSY, asterisk will return BUSY. Another way is to use the GROUPCOUNT set of functions, to keep a state in Asterisk or to use the embedded call counter in the SIP channel driver, that is reported in the SIPPEER function. As often is the case, there are many ways to solve an issue in Asterisk. /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
[asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? Any help? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] Telephone line signaling configuration in Egypt for FXO ports
Hi Bilal could you post the TDM configuration file (zaptel.conf and zapata.conf) and how did you compile it Regards Ayman Date: Wed, 13 Feb 2008 04:35:43 -0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? Any help? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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 _ Helping your favorite cause is as easy as instant messaging. You IM, we give. http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join___ -- 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] Telephone line signaling configuration in Egypt for FXO ports
On Wed, Feb 13, 2008 at 04:35:43AM -0800, bilal ghayyad wrote: Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? What version of zaptel? What do you have in zaptel.conf? -- 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] How to detect if SIP extension BUSY?
On 13:14, Wed 13 Feb 08, Gergo Csibra wrote: Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote: My problem is in subject. As I read in documentations and voip-info.org I can't user ChanIsAvalil because it not detects BUSY information on SIP channel. I've tried to use SIPPEER function, but it gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP extensions are Linksys PAP2. Have any other idea? Well? Is it impossible to detect BUSY on SIP channels? not in stock 1.2 Bristuff has a function for it, and russell created a function for it in current trunk that is also available as patch to 1.4 So you have 2 possibilities: - install bristuff 1.2 or 1.4 - install 1.4 with russell's patch applied bristuff examples: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate asterisk + function from russell: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] [Linux/Python 2.4.2] Forking Python doesn't work
Vincent,
try to use System() instead of AGI()
Diego Aguirre
Infodag - Informática
FWD#: 459696
Nikotel#: 99 21 8138-2710
EnumLookup#: +55 21 8138-2710
DUNDi-br#: 21 8138-2710
Vincent escreveu:
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned: It waits until the
script ends before moving on to the next step, which is Dial():
===
exten = s,1,AGI(netcid.py|${CALLERID(num)}|${CALLERID(name)}) exten
= s,n,Dial(${MYPHONE},5)
===
# cat netcid.py
#!/usr/bin/python
import socket,sys,time,os
def sendstuff(data):
s.sendto(data,(ipaddr,portnum))
return
sys.stdout = open(os.devnull, 'w')
if os.fork():
#BAD? sys.exit(0)
os._exit(0)
else:
now = time.localtime(time.time())
dateandtime = time.strftime(NaVm/%y NaVM, now)
myarray = []
myarray.append(STAT Rings: 1)
myarray.append(RING)
myarray.append(NAME + cidname)
myarray.append(TTSN Call from + cidname)
myarray.append(NMBR + cidnum)
myarray.append(TYPE K)
s = socket.socket(socket.AF_INET,socket.SOCK_DGRAM)
s.setsockopt(socket.SOL_SOCKET,socket.SO_BROADCAST,True)
portnum = 42685
ipaddr = 192.168.0.255
for i in myarray:
sendstuff(i)
#Must pause, and send IDLE for dialog box to close
time.sleep(5)
sendstuff(IDLE + dateandtime)
===
In another forum, people told me that I should fork twice. Is that
really necessary?
http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/278731
Thank you.
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Re: [asterisk-users] Attendant phone
voip crazy wrote: Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. I'd suggest looking at FOP (Flash Operator Panel). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
On 13:46, Wed 13 Feb 08, Vincent wrote: Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): In another forum, people told me that I should fork twice. Is that really necessary? http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/278731 If you want it to detach the program from it's parent you need the double fork yes. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] restart asterisk daily
On Wed, Feb 13, 2008 at 02:31:11PM +0100, randulo wrote: On Feb 13, 2008 9:29 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! Do I detect a tiny bit of sarcasm here? Someone from Digium (or elsewhere) might be able to jump in and explain the asterisk memory strategy and why it doesn't have any detrimental effects on anything else running on the same system. Sarcastic indeed. Indeed all those assertions were false. Off-Topic: The big memory consumer I have on my system is $GECKO_BROWSER. I currently have iceape (seamonkey), after just one day of operation: tzafrir 8186 1.1 53.1 763016 264008 ? Ssl Feb12 19:23 /usr/lib/iceape/iceape-bin Iceweasel (firefox), epiphany and kazehakase don't seem to be much different. So I have no issues with the little copy of Asterisk on my desktop system... -- 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
[asterisk-users] Wanted: VoIP Engineer for Switerland
Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemenshttp://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1sdc_sid=1113876080;and AVM/FRITZ!Box http://www.fritz-shop.ch/ . Due to our rapid growth, we are currently seeking for: VOIP SPECIALIST Place of work: Zurich *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - Experience with PHP - Practical knowledge of C and C++ - Practical knowledge of Mysql and Postgresql - Linux experience - Knowledge of IP Networks, UDP, TCP - Experience with tools for Network analysis like Ethereal - VoIP basic knowledge, VoIP servers, configuration of devices - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol, knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat would be an additional asset. Interested individuals are requested to send their resume to: [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] restart asterisk daily
On Feb 13, 2008 9:29 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! Do I detect a tiny bit of sarcasm here? Someone from Digium (or elsewhere) might be able to jump in and explain the asterisk memory strategy and why it doesn't have any detrimental effects on anything else running on the same system. ___ -- 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] UK issue - Asterisk dialling 999... sort of
Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. There are no phones connected in anyway other than via Asterisk, fax number is dealt with by a virtual machine, alarm system is on a different number... Any ideas before the rossers come and take me away? Phil Phil Knighton Support Engineer MJog Support Team Soft Option Technologies Ltd The Old School, 23 High Street, Wilburton, Cambridgeshire, CB6 3RB Tel: 01353 741641 | Fax: 01353 741341 | Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | Web: www.mjog.com http://www.mjog.com/ MJogClockOriginal4.jpg___ -- 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] restart asterisk daily
On Wed, Feb 13, 2008 at 03:02:23PM +0100, Haan Patrick wrote: which distribution do you use? Maybe a Fedora 7 Debian Testing here. -- 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
[asterisk-users] What is a secure call?
Friends,
The following mail was sent earlier to asterisk-dev and did not cause
the amount of discussion I hoped it would.
Now that we have a way to secure signalling in IAX2 and SIP in
Asterisk svn trunk, we need to start working on
the concept of a secure call - or does it really matter?
In SIP, there's a specification for how I as a domain owner can
request all calls to my domain to use
SIP/TLS by using DNS NAPTR and SRV records. But how do I as a caller
request a secure service?
How do we place a secure call with DIAL? Do we need SECUREDIAL?
Any ideas and thoughts on the subject are welcome!
Regards,
/Olle
- Copy of earlier mail -
(http://lists.digium.com/pipermail/asterisk-dev/2007-July/028377.html)
To open a can of worms... :-)
I'm involved in Phil Zimmerman's efforts to integrate Zrtp into
Asterisk. At the same time we have code for SRTP that needs to
be integrated.
This means that we will add the concept of a secure call in
Asterisk. At some point, I want to be able to build dialplans
where I can manager security requirements on channels, like this
conference is protected and requires a secure channel.
So, to make this easy, should we have a boolean flag and determine
this is a secure call according to Asterisk Community
Security Standards or how should we handle this? I think leaving it
up to the admin is the proper way to go, but we
also have several scenarios to consider
1. Encrypted signalling and media stream
1. Open signalling stream, key exchange in the open, encrypted media
2. Open signalling stream, secure key exchange, encrypted media
3. Secure signalling stream, un-encrypted media
exten = _x.,n,gotoif(${ISSECURECALL(level6)} ? approved,1 :
hangup,1)
And to add to that, we have many different call scenarios.
1. Bridged call between two secure endpoints, Asterisk transcodes and
have an unsecure media path
2. One-legged secure call between Asterisk and a phone (IVR)
3. SIP to ASterisk over IAX trunk to another Asterisk to SIP with SRTP/
TLS and encrypted IAX - but open
media path when going from SIP to IAX
And yes, of course, this is not attempting to be a complete list at all.
Can we simplify this and make it configurable? Do we want to?
Can we implement the notion of a trusted PBX that we allow being in
the middle and untrusted PBXs
that we want to avoid or that changes the security property of a call.
As I said to Phil: A PBX is designed to be a man-in-the-middle attack.
There's certainly room for discussion, brainstorming and wild ideas
here.
/O
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Re: [asterisk-users] urgent-channels
On Wed, 2008-02-13 at 11:33 +0200, Khaled Chehab wrote: I am using asterisk 1.2.4 Version 1.2.4 is really quite old (it was released in January of 2006, so is more than 24 months old at this point), and there have been hundreds of bugs fixed since then. I'd really suggest you try a newer version of Asterisk, either the 1.2.26.2 for the 1.2 branch, or 1.4.18 on the 1.4 branch. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] restart asterisk daily
On 2/13/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Feb 13, 2008 at 03:48:14PM +0800, Rilawich Ango wrote: Actually, I donno it is a memory leak or not. I have a server only running asterisk. As time goes by, the free memory shown in the top is decreased. After I restart the asterisk, the free memory comes again. That's why I wonder if regular restart asterisk is necessary. Use a crontab to restart asterisk is a way to do it but you have to maintain a crontab. Is it possible to use logrotate instead? Or other better way? [EMAIL PROTECTED]:~$ free -m total used free sharedbuffers cached Mem: 485477 7 0 0100 -/+ buffers/cache:376108 Swap: 1419270 1149 [EMAIL PROTECTED]:~$ top -b | head -n 5 top - 10:18:32 up 19 days, 14:38, 24 users, load average: 0.08, 0.33, 0.21 Tasks: 166 total, 1 running, 163 sleeping, 2 stopped, 0 zombie Cpu(s): 1.1%us, 0.1%sy, 0.0%ni, 98.2%id, 0.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem:496648k total, 489044k used, 7604k free, 32k buffers Swap: 1453840k total, 276740k used, 1177100k free, 103380k cached [EMAIL PROTECTED]:~$ ps aux | grep asterisk asterisk 9559 0.0 2.5 474896 12892 ?Ssl Feb12 0:00 /usr/sbin/asterisk -p -U asterisk Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! Guys, don't start panic here. This is perfectly normal memory status for Linux. Linux automatically uses most free memory for disk cache, leaving only few megabytes, and frees disk cache as soon as any application requests. This has nothing to do with Asterisk. Regards, Atis In fact: 1. The system has some 100MB of free memory. almost all of it is used for caching and such. 2. Asterisk overcommits memory: it generally asks the kernel huge ammounts of memory, but doesn't really try to use them. At least with Linux such overcommits are not claimed at all. -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] restart asterisk daily
So that´s why I´ve always get a red bar on home screen of the Trixbox? Phisical memory is always at top most use, near 100% (green bar turns red on high level of memory use), and below it there is Kernel / Application, Buffers, Cached memory uses. tks, On Feb 13, 2008 12:51 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On 2/13/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Feb 13, 2008 at 03:48:14PM +0800, Rilawich Ango wrote: Actually, I donno it is a memory leak or not. I have a server only running asterisk. As time goes by, the free memory shown in the top is decreased. After I restart the asterisk, the free memory comes again. That's why I wonder if regular restart asterisk is necessary. Use a crontab to restart asterisk is a way to do it but you have to maintain a crontab. Is it possible to use logrotate instead? Or other better way? [EMAIL PROTECTED]:~$ free -m total used free sharedbuffers cached Mem: 485477 7 0 0 100 -/+ buffers/cache:376108 Swap: 1419270 1149 [EMAIL PROTECTED]:~$ top -b | head -n 5 top - 10:18:32 up 19 days, 14:38, 24 users, load average: 0.08, 0.33, 0.21 Tasks: 166 total, 1 running, 163 sleeping, 2 stopped, 0 zombie Cpu(s): 1.1%us, 0.1%sy, 0.0%ni, 98.2%id, 0.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem:496648k total, 489044k used, 7604k free, 32k buffers Swap: 1453840k total, 276740k used, 1177100k free, 103380k cached [EMAIL PROTECTED]:~$ ps aux | grep asterisk asterisk 9559 0.0 2.5 474896 12892 ?Ssl Feb12 0:00 /usr/sbin/asterisk -p -U asterisk Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! Guys, don't start panic here. This is perfectly normal memory status for Linux. Linux automatically uses most free memory for disk cache, leaving only few megabytes, and frees disk cache as soon as any application requests. This has nothing to do with Asterisk. Regards, Atis In fact: 1. The system has some 100MB of free memory. almost all of it is used for caching and such. 2. Asterisk overcommits memory: it generally asks the kernel huge ammounts of memory, but doesn't really try to use them. At least with Linux such overcommits are not claimed at all. -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] restart asterisk daily
which distribution do you use? Maybe a Fedora 7 greez patrick -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Tzafrir Cohen Gesendet: Mittwoch, 13. Februar 2008 14:46 An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] restart asterisk daily [senderbase] On Wed, Feb 13, 2008 at 02:31:11PM +0100, randulo wrote: On Feb 13, 2008 9:29 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Gee, I only have 7 MB free! I must reboot to free some memory! And that Asterisk is using so much memory! Do I detect a tiny bit of sarcasm here? Someone from Digium (or elsewhere) might be able to jump in and explain the asterisk memory strategy and why it doesn't have any detrimental effects on anything else running on the same system. Sarcastic indeed. Indeed all those assertions were false. Off-Topic: The big memory consumer I have on my system is $GECKO_BROWSER. I currently have iceape (seamonkey), after just one day of operation: tzafrir 8186 1.1 53.1 763016 264008 ? Ssl Feb12 19:23 /usr/lib/iceape/iceape-bin Iceweasel (firefox), epiphany and kazehakase don't seem to be much different. So I have no issues with the little copy of Asterisk on my desktop system... -- 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 ___ -- 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] LCR in Asterisk
On Tuesday 12 February 2008 23:14:58 Alex Balashov wrote: Rizwan Hisham wrote: Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which can be used with asterisk. If you are interested in prebuilt solutions, you may consider TransNexus's NexOSS product (www.transnexus.com). The Open Settlement Protocol (OSP) they implemented can be used with Asterisk - they have a module. In fact, I am not sure about the commercial status of the OSP module as such; it may be possible to get it free of charge. Not sure. But OSP is an open protocol, so perhaps it's possible. Otherwise, I would think that the best way to approach this would be to make it fully outboard and divest it of Asterisk. Implement a SIP proxy that forwards to providers using LCR decisionmaking, and just have Asterisk send calls to it. OpenSER can be used for this - and indeed, there is an OSP module for it as well, if you wanted to go that route. If you're dead-set on doing it in Asterisk and don't want to do OSP, I would suggest FastAGI. Definitely don't implement the logic in the dial plan, at any cost. Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. -- 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] UK issue - Asterisk dialling 999... sort of
It might be possible to get to the emergency service via 112 or a local number as well. Do you have *any* calls placed at about the time of the 999 calls? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Knighton Sent: 13 February 2008 14:12 Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. There are no phones connected in anyway other than via Asterisk, fax number is dealt with by a virtual machine, alarm system is on a different number... Any ideas before the rossers come and take me away? Phil http://www.mjog.com/ ___ -- 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] Digium's Exceptional Satisfaction Program
As many of you may well know, Digium has been investing a great deal of time and effort to build the very best telephony products in the industry. We're committed to producing the highest quality hardware and software solutions, along with things like training and support to make your Asterisk deployment a successful one. As part of this effort, Digium is launching its Exceptional Satisfaction Program. I won't bore you with all of the details here (see links below for more detailed info), but in a nutshell we've extended the warranties on almost our entire line of hardware and commercial software products, and have thrown in a money-back guarantee as well. The blog post announcing the program can be found at http://blogs.digium.com/2008/02/11/digium-puts-its-money-where-its-mouth-is/. The details of the program can be found at http://www.digium.com/ESP. We've also created a FAQ page at http://www.digium.com/en/company/riskfree-facts.php. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] is encrypted iax safe and secure?
[EMAIL PROTECTED] wrote: Is it important for you to conceal that a call was made from abc to xyz on thus-and-such a date? Or do you merely need to conceal the content of a call? I was thinking about concealing called and calling number in a generic iax2 call, I hadn't even thinked about concealing the call itself. :-) Another not so related question, during iax2 registration is username Information Element always sent in clear? I guess it is in clear since the first REGREQ even in the case of RSA or MD5 based authentication. Thanks, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- 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] UK issue - Asterisk dialling 999... sort of
On Wednesday 13 February 2008 08:12:25 Phil Knighton wrote: Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. Have you checked all numbers that might have a PREFIX of 999? Here in the States, occasionally a prankster will tell someone annoying her to call her on her cell phone at 911-5924 or something like that, and of course, the system only sees the 911 portion, not the additional 4 digits, which connects them to the emergency number on this side of the Pond. -- 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] GXP2000 and asterisk 1.0.9
Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- 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] FOSDEM in Brussells - Feb 23-24
Friends, I will be attending FOSDEM in Brussells Feb 23-24. Anyone else? Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be there, so we could have a SIP/XMPP/Asterisk ad hoc meeting :-) On Thursday, Feb 21, I will be in Utrecht, Netherlands for the free Open Telephony conference at Media Plaza. There's still seats available and a really good talk about ENUM with Patrik Fältström, Cisco/IETF. Join us there! Register at http://www.mediaplaza.nl/mp.php/mediaplaza/agenda/agenda.php?id=312 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
Re: [asterisk-users] LCR in Asterisk
Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. It's not syntactically terse, balloons quickly in semantic complexity, is objectively less efficient as the dial plan *is not a programming language* (despite having variables, control structures and other things characteristic of an execution environment of such), and otherwise unnecessarily complicated. In implementing and extending the logic going forward (beyond naive lookups) in accordance with evolving requirements in the business rules, you will find that you run into the limits of the algorithmic complexity that the dial plan can provide, and that whatever the approach, it's overly obfuscated. The dial plan limits meaningful modularisation and functional decomposition that is available with outboard runtime environments. So, it's not that you couldn't - it's that you shouldn't. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Analog DID
Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 ___ -- 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] Analog DID
On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- 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] Asterisk and fax
Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Asterisk and fax
Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is a secure call?
If Asterisk does indeed use SECUREDIAL or similar as distinct from
DIAL, then DIAL should wrap SECUREDIAL for calls to a party that are
secure. This would parallel HTTP GET (or POST) which use the same
function entry for both secure and insecure connections, wrapping their
secure access inside generic access.
To continue the parallel, the dialstring should indicate whether
SIP/TLS (and otherwise for IAX) is to be used, which should allow the
DIAL function to determine whether to make a secure connection. To go
further, if SECUREDIAL is invoked on a dialstring which does not specify
a secure connection, that invocation should at least flag the insecure
connection attempt, or even fail with an exception.
I'm not sure that the SIP spec allows a request for an insecure
connection to be rejected with instructions for requesting a secure
call. But if it does, then the DIAL function should allow logic for
options on the retry, like just failing with exception report or a list
of dialstrings to retry. Or maybe just an extention to jump to with the
failure in a variable, for the dialplan/AGI/etc able to use that status
for logic on retry or fail.
In general, the closer the DIAL function works to familiar Web
retrieval functions, the more Web programming techniques will be
applicable to Asterisk programming.
On Wed, 2008-02-13 at 10:40 -0600,
[EMAIL PROTECTED] wrote:
Date: Wed, 13 Feb 2008 15:22:10 +0100
From: Johansson Olle E [EMAIL PROTECTED]
Subject: [asterisk-users] What is a secure call?
To: Asterisk Non-Commercial Discussion Users Mailing List -
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
Friends,
The following mail was sent earlier to asterisk-dev and did not
cause
the amount of discussion I hoped it would.
Now that we have a way to secure signalling in IAX2 and SIP in
Asterisk svn trunk, we need to start working on
the concept of a secure call - or does it really matter?
In SIP, there's a specification for how I as a domain owner can
request all calls to my domain to use
SIP/TLS by using DNS NAPTR and SRV records. But how do I as a caller
request a secure service?
How do we place a secure call with DIAL? Do we need SECUREDIAL?
Any ideas and thoughts on the subject are welcome!
Regards,
/Olle
- Copy of earlier mail -
(http://lists.digium.com/pipermail/asterisk-dev/2007-July/028377.html)
To open a can of worms... :-)
I'm involved in Phil Zimmerman's efforts to integrate Zrtp into
Asterisk. At the same time we have code for SRTP that needs to
be integrated.
This means that we will add the concept of a secure call in
Asterisk. At some point, I want to be able to build dialplans
where I can manager security requirements on channels, like this
conference is protected and requires a secure channel.
So, to make this easy, should we have a boolean flag and determine
this is a secure call according to Asterisk Community
Security Standards or how should we handle this? I think leaving
it
up to the admin is the proper way to go, but we
also have several scenarios to consider
1. Encrypted signalling and media stream
1. Open signalling stream, key exchange in the open, encrypted media
2. Open signalling stream, secure key exchange, encrypted media
3. Secure signalling stream, un-encrypted media
exten = _x.,n,gotoif(${ISSECURECALL(level6)} ? approved,1 :
hangup,1)
And to add to that, we have many different call scenarios.
1. Bridged call between two secure endpoints, Asterisk transcodes
and
have an unsecure media path
2. One-legged secure call between Asterisk and a phone (IVR)
3. SIP to ASterisk over IAX trunk to another Asterisk to SIP with
SRTP/
TLS and encrypted IAX - but open
media path when going from SIP to IAX
And yes, of course, this is not attempting to be a complete list at
all.
Can we simplify this and make it configurable? Do we want to?
Can we implement the notion of a trusted PBX that we allow being
in
the middle and untrusted PBXs
that we want to avoid or that changes the security property of a call.
As I said to Phil: A PBX is designed to be a man-in-the-middle
attack.
There's certainly room for discussion, brainstorming and wild ideas
here.
/O
--
(C) Matthew Rubenstein
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Re: [asterisk-users] Analog DID
An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price.Darren Wiebe[EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DIDOn Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks?What is an analog DID trunk?You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500This is for providing plenty of analog extensions (phones). Is that whatyou're after?-- Tzafrir Cohenicq#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 listTo 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] FOSDEM in Brussells - Feb 23-24
On 16:59, Wed 13 Feb 08, Johansson Olle E wrote: Friends, I will be attending FOSDEM in Brussells Feb 23-24. Anyone else? I'll be there (what a suprise eh ?) Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be there, so we could have a SIP/XMPP/Asterisk ad hoc meeting :-) yeah, we should meet and checkout the supply of Gulden Draak On Thursday, Feb 21, I will be in Utrecht, Netherlands for the free Open Telephony conference at Media Plaza. There's still seats available and a really good talk about ENUM with Patrik F?ltstr?m, Cisco/IETF. Join us there! I'll be there as well. Register at http://www.mediaplaza.nl/mp.php/mediaplaza/agenda/agenda.php?id=312 meh, I was registered before I knew ;) Regards, /Olle -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] LCR in Asterisk
On Wed, Feb 13, 2008 at 11:33:19AM -0600, Tilghman Lesher wrote: On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote: Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. It's not syntactically terse, balloons quickly in semantic complexity, is objectively less efficient as the dial plan *is not a programming language* (despite having variables, control structures and other things characteristic of an execution environment of such), and otherwise unnecessarily complicated. In implementing and extending the logic going forward (beyond naive lookups) in accordance with evolving requirements in the business rules, you will find that you run into the limits of the algorithmic complexity that the dial plan can provide, and that whatever the approach, it's overly obfuscated. The dial plan limits meaningful modularisation and functional decomposition that is available with outboard runtime environments. Like any other language, you certainly can write in an obfuscated way, and the dialplan does not discourage it. That said, you certainly can write in a modularized way. I would guess that you simply aren't familiar with the dialplan enough to make those decisions, but it is quite possible and doable. So, it's not that you couldn't - it's that you shouldn't. In the same way that a PHP programmer should not attempt write Python the way she writes PHP, I would agree with you. However, if you're willing to adapt to the ways the dialplan works, you can create dialplans which aren't obfuscated at all. Tcl and Lisp are close cousins to the dialplan in terms of how they do things. Not everybody is a Lisp programmer, and some people absolutely detest it. That doesn't make it any less of a good language. Having programmed in about 8 different languages over the last 25 years, I can see both points of view. And admittedly, I haven't tried to do non-trivial things with dialplan. That said, my view of this interaction is that Tilghman has drunk the Kool-Aidtm, and that Alex's view of the situation is much closer to objective. dialplan appears to have jes' growed, and that never makes for a good language design. Ask the Python 3 team. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] LCR in Asterisk
On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote: Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. It's not syntactically terse, balloons quickly in semantic complexity, is objectively less efficient as the dial plan *is not a programming language* (despite having variables, control structures and other things characteristic of an execution environment of such), and otherwise unnecessarily complicated. In implementing and extending the logic going forward (beyond naive lookups) in accordance with evolving requirements in the business rules, you will find that you run into the limits of the algorithmic complexity that the dial plan can provide, and that whatever the approach, it's overly obfuscated. The dial plan limits meaningful modularisation and functional decomposition that is available with outboard runtime environments. Like any other language, you certainly can write in an obfuscated way, and the dialplan does not discourage it. That said, you certainly can write in a modularized way. I would guess that you simply aren't familiar with the dialplan enough to make those decisions, but it is quite possible and doable. So, it's not that you couldn't - it's that you shouldn't. In the same way that a PHP programmer should not attempt write Python the way she writes PHP, I would agree with you. However, if you're willing to adapt to the ways the dialplan works, you can create dialplans which aren't obfuscated at all. Tcl and Lisp are close cousins to the dialplan in terms of how they do things. Not everybody is a Lisp programmer, and some people absolutely detest it. That doesn't make it any less of a good language. -- 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] Analog DID
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino's Analog cards support analog DID. no need for all the extra stuff You will want to get an R8FXX with fxs modules that will give you channels in sets of 2. ADID has not really taken off in the OS telephony market I think due to a lack of understanding people stay with the proprietary phone systems that pimp this feature. Okay so I will take the lead and pimp it for asterisk. With Rhino Analog cards you CAN do ADID with no extra equipment. However if you want to spend the money we can go the other route :) darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- 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 !DSPAM:47b327c1163231152562594! -- ___ -- 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 !DSPAM:47b327c1163231152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKB Gxd6H7YOdzXfygVuBygzAw== =51QY -END PGP 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] Asterisk and fax
I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] ISDN PRIs and taking a server down for maintenance - blocking issue
Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What I would like to know is if there is anyway to get around this and not send a BUSY back to our callers and somehow allow our telco to present calls immediately to SERVER2. Anyone have any ideas or are we stuck with this behaviour until the calls drop to 0 and Asterisk shuts down ? Thanks, Andrew ___ -- 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] ISDN PRIs and taking a server down for maintenance - blocking issue
Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Andrew Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What I would like to know is if there is anyway to get around this and not send a BUSY back to our callers and somehow allow our telco to present calls immediately to SERVER2. Anyone have any ideas or are we stuck with this behaviour until the calls drop to 0 and Asterisk shuts down ? Thanks, Andrew ___ -- 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] LCR in Asterisk
- Original Message From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 9:45:34 AM Subject: Re: [asterisk-users] LCR in Asterisk On Wed, Feb 13, 2008 at 11:33:19AM -0600, Tilghman Lesher wrote: On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote: Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. Is that nasty little problem of no local variables in macros fixed yet? That's a pretty big pain in the ass. You have to prefix your variables with the name of the macro it's in to avoid stepping all over yourself. Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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 and fax
I'm at this moment implementing the same as you do... Take a look at the following links: http://blog.evaristesys.com/?p=24 http://blogtech.oc9.com/index.php?option=com_contentview=articlecatid=4:asteriskid=77:20071121astItemid=6 http://www.voip-info.org/wiki/view/Asterisk+fax Regards, Ricardo Carvalho. On Feb 13, 2008 5:49 PM, voip crazy [EMAIL PROTECTED] wrote: I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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 ___ -- 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] LCR in Asterisk
Douglas Garstang wrote: - Original Message From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 9:45:34 AM Subject: Re: [asterisk-users] LCR in Asterisk On Wed, Feb 13, 2008 at 11:33:19AM -0600, Tilghman Lesher wrote: On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote: Tilghman Lesher wrote: Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc for each of the provider lookups, then use SORT() to get the lowest cost. It's quite easy, and you do not need to resort to AGI. You can do almost anything in the dial plan with enough spiritual commitment in about the same way that you can do just about anything you need to do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. Could you fix your e-mail client please? Regards, Philipp Kempgen ___ -- 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] LCR in Asterisk
On Wednesday 13 February 2008 11:45:34 Jay R. Ashworth wrote: Having programmed in about 8 different languages over the last 25 years, I can see both points of view. And admittedly, I haven't tried to do non-trivial things with dialplan. That said, my view of this interaction is that Tilghman has drunk the Kool-Aidtm, and that Alex's view of the situation is much closer to objective. Or maybe I'm just the architect of the dialplan moving forward, which is why I advocate that if you really don't need to use AGI, you don't. ;-) dialplan appears to have jes' growed, and that never makes for a good language design. Ask the Python 3 team. :-) I'm specifically working on removing misfeatures from the dialplan, to make it much easier to use and more predictable. 1.6 will be a huge improvement towards this goal. -- 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] MWI problem with Siemens Gigaset S675 IP
Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. Thanks, Jaap ___ -- 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] LCR in Asterisk
At 09:33 AM 2/13/2008, you wrote: In the same way that a PHP programmer should not attempt write Python the way she writes PHP, I would agree with you. However, if you're willing to adapt to the ways the dialplan works, you can create dialplans which aren't obfuscated at all. Tcl and Lisp are close cousins to the dialplan in terms of how they do things. Not everybody is a Lisp programmer, and some people absolutely detest it. That doesn't make it any less of a good language. Look, I've done lots of cool stuff in the dial plan and other have done stuff way beyond me, but I defy you to call the dial plan language good or well designed. It works, it gets the job done but it's always harder than it needs to be. 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] Analog DID
Hey, that's cool! I wish I'd known that 6 months ago.Darren Wiebe[EMAIL PROTECTED]Wed Feb 13 2008 10:33:31 AM MST from James Finstrom to Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Analog DID-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Rhino's Analog cards support analog DID. no need for all the extrastuff You will want to get an R8FXX with fxs modules that will giveyou channels in sets of 2.ADID has not really taken off in the OS telephony market I think dueto a lack of understanding people stay with the proprietary phonesystems that pimp this feature. Okay so I will take the lead and pimpit for asterisk. With Rhino Analog cards you CAN do ADID with no extraequipment. However if you want to spend the money we can go the otherroute :)darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- 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 !DSPAM:47b327c1163231152562594! -- ___ -- 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 !DSPAM:47b327c1163231152562594!- --James FinstromRhino Equipment Corp.Tel: 1-800-785-7073 ext. 6344FAX: +1 (480) 961-1826IP: asterisk.rhinoequipment.com ext 6344FWD: 633686 ext 6344THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARYMATERIAL and is thus for use only by the intended recipient. If youreceivedthis in error, please contact the sender and delete the email and itsattachments from all computers.-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.6 (GNU/Linux)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKBGxd6H7YOdzXfygVuBygzAw===51QY-END PGP SIGNATURE-___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo 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] Asterisk and fax
I would recommend you use Iaxmodem / Hylafax / Avantfax for your needs. We use this with several customers and it works very well. This way you do not have to patch Asterisk with spanDSP. You can set up as many virtual fax machines as your machine will handle. On Wed, 2008-02-13 at 18:49 +0100, voip crazy wrote: I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] LCR in Asterisk
Doug- Please fix your email client. One line per word in quoting is a little excessive. Better yet, turn off HTML. On Wednesday 13 February 2008 12:17:30 Douglas Garstang wrote: Is that nasty little problem of no local variables in macros fixed yet? That's a pretty big pain in the ass. You have to prefix your variables with the name of the macro it's in to avoid stepping all over yourself. Macros are deprecated. Gosubs are the way forward, and yes, they have local variables. Simply define them once as Set(LOCAL(foo)=bar) and foo will be gone when the innermost stack is removed (either by Return or StackPop). -- 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] LCR in Asterisk
On Wed, Feb 13, 2008 at 12:52:42PM -0600, Tilghman Lesher wrote: On Wednesday 13 February 2008 11:45:34 Jay R. Ashworth wrote: Having programmed in about 8 different languages over the last 25 years, I can see both points of view. And admittedly, I haven't tried to do non-trivial things with dialplan. That said, my view of this interaction is that Tilghman has drunk the Kool-Aidtm, and that Alex's view of the situation is much closer to objective. Or maybe I'm just the architect of the dialplan moving forward, which is why I advocate that if you really don't need to use AGI, you don't. ;-) I'm not sure those aren't equivalent. :-) dialplan appears to have jes' growed, and that never makes for a good language design. Ask the Python 3 team. :-) I'm specifically working on removing misfeatures from the dialplan, to make it much easier to use and more predictable. 1.6 will be a huge improvement towards this goal. Well, this should be interesting. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] LCR in Asterisk
On Wed, Feb 13, 2008 at 07:49:36PM +0100, Philipp Kempgen wrote: Douglas Garstang wrote: [ ... ] do with a bash script, as opposed to Perl, Python, or any toolkits or frameworks. Could you fix your e-mail client please? I dunno; his message comes out fine here, though Mutt and lynx --dump. I grow less impressed with T-bird by the day... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] LCR in Asterisk
Tilghman Lesher wrote: Like any other language, you certainly can write in an obfuscated way, and the dialplan does not discourage it. That said, you certainly can write in a modularized way. I would guess that you simply aren't familiar with the dialplan enough to make those decisions, but it is quite possible and doable. The dial plan certainly does lend itself to this to some degree, no argument, but not to the extent that fully developed programming / scripting languages do. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] PCI32 and PCI-X compatibility
this is my 1st message, I'm writing to ask if anyone knows if a PCI32 card like the TDM400P (quad analog) or the B410P (quad BRI) is working on a PCI-X bus, at 100MHz or 133 MHz. I'm really stuck with this, since I found a partial yes on this mailing list but my supplier says no! Marco, You should not have any issues using a PCI card in a PCI-X slot, as long as the card is a 3.3V PCI card. The cards that you mention above are 3.3v compatible and you should be able to use them. All of Digium's product line is available for 3.3v slots. Most are universal and can be used in 3.3v or 5v slots. The only exceptions are the dual and quad span T1/E1 digital cards. For those cards, there are 3.3v variants (TE410P and TE210P) and 5v variants (TE405P and TE410P). -- Michael Spiceland ___ -- 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] GXP2000 and asterisk 1.0.9
Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- 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] Asterisk Manager and Visual Basic
Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? I've got a Telnet control that is allowing me to connect, authenticate and see the flow of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any suggestions? I want to write a presence monitor (a virtual sidecar if you will) Bill ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre [EMAIL PROTECTED] wrote: try to use System() instead of AGI() Thanks, but no go. I get an error: [Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper: Unable to execute '/tmp/netcid.py|2000|Joe' ___ -- 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] MWI problem with Siemens Gigaset S675 IP
Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is) I've had to add the voicemail context to get MWI to work correctly in the past. - Original Message - From: Jaap Winius [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:45 PM Subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. Thanks, Jaap ___ -- 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] GXP2000 and asterisk 1.0.9
Is your phone actually registered to the switch. go to the CLI and do a 'sip show peers' see if extension 502 is registered. Making an outbound call does not prove that the phone is registered. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 2:09 PM Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- 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 ___ -- 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 Manager and Visual Basic
On 13/02/2008, Bill Andersen [EMAIL PROTECTED] wrote: Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? Bill I wrote some very very basic stuff ages ago using standard mswinsck.ocx, will dig it out. ___ -- 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] urgent-channels
Khaled Chehab wrote: Hi All I am using asterisk 1.2.4 Please see the results when I execute Sip show channels *569 *active SIP channels What phones are you using? We had a similar problem with Snom 360 phones with firmware version 6.2.2 and asterisk 1.2, whereby channels would not hangup correctly. Cheers, Ben ___ -- 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] UK issue - Asterisk dialling 999... sort of
When I first set up asterisk, I had similar, fortunately not with the old bill! It basically boiled down to not enough delay between seizing the line and dialing the digits, for example the following would have dialled the coppers 012*99 9*12345, as 012 would have been missed. I'm guessing this isn't whats happening to you, if all your other calls are uworking fine, but did bring back some memories and made me smile :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] Asterisk and fax
I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). We use (at multiple sites, mostly BRI) iaxmodem and hylafax. Extra bonus: you get all the cool features and possibilities of hylafax! ;-) cheers, stoffell ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
On Wed, 13 Feb 2008 14:25:52 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
If you want it to detach the program from it's parent you
need the double fork yes.
Thanks for the confirmation, but when doing this, the NetCID
application no longer pops up, regardless of whether I put the NetCID
code in the second parent or second child:
exten = 9300,1,AGI(/tmp/test5.py|${CALLERID(num)}|${CALLERID(name)})
exten = 9300,n,Dial(${MYPHONE},15)
# cat test5.py
#!/usr/bin/python
import socket,sys,time,os
def sendstuff(data):
s.sendto(data,(ipaddr,portnum))
return
log = open('/tmp/output.txt','w')
sys.stdout = open(os.devnull, 'w')
if os.fork():
#Parent
log.write(Step 1\n)
log.close()
os._exit(0)
else:
#Child
os.chdir('/tmp')
os.setsid()
os.umask(0)
if os.fork():
#Parent
log.write(Step 2\n)
log.close()
now = time.localtime(time.time())
dateandtime = time.strftime(%d/%m/%Y %H:%M, now)
myarray = []
myarray.append(STAT Rings: 1)
myarray.append(RING)
myarray.append(NAME + cidname)
myarray.append(TTSN Call from + cidname)
myarray.append(NMBR + cidnum)
myarray.append(TYPE K)
s = socket.socket(socket.AF_INET,socket.SOCK_DGRAM)
s.setsockopt(socket.SOL_SOCKET,socket.SO_BROADCAST,True)
portnum = 42685
ipaddr = 192.168.0.255
for i in myarray:
sendstuff(i)
time.sleep(5)
sendstuff(IDLE + dateandtime)
os._exit(0)
else:
#Child
log.write(Step 3\n)
log.close()
os._exit(0)
Has someone already forked a Python script successfully from Asterisk
through AGI?
Thanks.
___
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Re: [asterisk-users] Realtime SIP peers - reloading cached info
Johansson Olle E wrote:
So please rememner that there are a few independent regular Asterisk
developers out there that is not on the Digium payroll and still take
part in decisions about Asterisk.
Point taken.
Over a year is a long time for a warning like this, considering that
1.6 won't be out for a while (we're in beta test cycle) it might even
be 1.5 year warning. That should be more than enough for most people -
I hope. Considering that people don't upgrade quickly, it will
propably be more than that for most users (as you are still on 1.2 :-) )
You could be right there - though my main concern is that since I'm
developing for /mostly/ 1.2 systems at this stage, I can't use the new
syntax since (as far as I can tell) the ${REALTIME} function isn't
available in 1.2. If it were, I'd convert my scripts /now/.
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Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Quoting Henry Devito [EMAIL PROTECTED]: Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is) I've had to add the voicemail context to get MWI to work correctly in the past. According to the documentation, you shouldn't have to add @context if the context is 'default'. But, I went ahead and tried it out anyway. I even tried using some other context names, but it makes no difference: the error remains the same. Thanks anyway, Jaap ___ -- 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] SIP over TCP
I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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] multiple host in 1 context on sip.conf
Is it possilble for a single context to have multiple host= something like this [carrier] host=ip address1 host=ip address2 host=ip address3 type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes -- Regards, Mark Quitoriano http://asterisk.org.ph Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Looks like it is part of the 1.6 Beta. From the Change Log: 2008-01-18 22:04 + [r99080-99085] Russell Bryant [EMAIL PROTECTED] * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) On Feb 13, 2008 4:21 PM, Razza [EMAIL PROTECTED] wrote: I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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] PCI32 and PCI-X compatibility
Marco, You should not have any issues using a PCI card in a PCI-X slot, as long as the card is a 3.3V PCI card. The cards that you mention above are 3.3v compatible and you should be able to use them. All of Digium's product line is available for 3.3v slots. Most are universal and can be used in 3.3v or 5v slots. The only exceptions are the dual and quad span T1/E1 digital cards. For those cards, there are 3.3v variants (TE410P and TE210P) and 5v variants (TE405P and TE410P). Oops, I meant that the 5v variants are the TE405P and *TE205P*. 3.3v - TE410P and TE210P 5v - TE405P and TE205P Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
SIP over TCP is included in 1.6. http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co On Feb 13, 2008 5:21 PM, Razza [EMAIL PROTECTED] wrote: I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. ___ -- 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 -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- 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] message: !! Got Busy in Connected State !?!
What phone do you use? Linksys ? Vieri schreef: --- Fons van der Beek [EMAIL PROTECTED] wrote: Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call off hold And then the external line is disconnected. I get the same with Asterisk 1.2 and chan_misdn. Is this a known bug or something I misconfigured? In the latter case, what should I look for? Thanks, Vieri Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] Attendant phone
As far as I'm aware, only the Aastra 57i with three 560M modules would come close to your requirements. The 57i can display up to 5 extensions at one time with a further 15 being available by the use of multiple pages. The 560M modules can display up to 20 extensions at one time with three pages being available for a total of 60 extensions per phone. This gives you a total of 200 extensions that can be monitored. voip crazy wrote: Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? ___ -- 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] Touch monitor file name format
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands
after the somix sequence for mp3 conversion. This should work, but
I've so far failed to produce any mp3 files because I'm not able to
predict the above epoch number. If I could alter 'auto-${EPOCH}-', or
if it was stored in a variable I could use, then I'm sure my plan will
succeed.
Any ideas?
Thanks,
Jaap
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Re: [asterisk-users] Attendant phone
To me it sounds like you should be using the Flash Operator Panel to monitor that many extensions. The Polycom 6xx range can monitor 42 extensions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Thursday, 14 February 2008 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Attendant phone As far as I'm aware, only the Aastra 57i with three 560M modules would come close to your requirements. The 57i can display up to 5 extensions at one time with a further 15 being available by the use of multiple pages. The 560M modules can display up to 20 extensions at one time with three pages being available for a total of 60 extensions per phone. This gives you a total of 200 extensions that can be monitored. voip crazy wrote: Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? ___ -- 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] [Linux/Python 2.4.2] Forking Python doesn't work
Vincent wrote: On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre [EMAIL PROTECTED] wrote: try to use System() instead of AGI() Thanks, but no go. I get an error: [Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper: Unable to execute '/tmp/netcid.py|2000|Joe' The arguments to System() are a bit different. Put it in just like you would type at the command line. System(/tmp/netcid.py 2000 Joe) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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 Manager and Visual Basic
- Original Message From: Bill Andersen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 13 February, 2008 8:31:01 PM Subject: [asterisk-users] Asterisk Manager and Visual Basic Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? I've got a Telnet control that is allowing me to connect, authenticate and see the flow of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any suggestions? Hi Bill, I don't know if it would be of any use to you but we have some C# code that handles the basics of communicating the the Asterisk Manager Interface. It doesn't do anything fancy just sends single commands and checks the responses. We don't use it for monitoring. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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
