Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Thanks a lot guys for your answers. I'll go ahead with NI-2. I didn't know its the same thing as National. Thanks again, Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com On 2011-01-21 10:36 AM, Bruce B bruceb...@gmail.com wrote: Yes, it does. Bell provides the same as well and it works with Asterisk. -Bruce On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer ... -- _ -- Bandwidth and Colo... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is equivalent function to mv command in php for Asterisk Spool directory usage?
I have been using:
exec ('mv *.call /var/spool/asterisk/outgoing')
and for a long time it has been working just fine for me on more than one
websites. Just make sure the folder where you create the call files has
correct permissions and ownerships so that the file is successfully moved by
the apache user to its destination.
Zeeshan A Zakaria
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On 2010-12-21 3:29 PM, Danny Nicholas da...@debsinc.com wrote:
PERL has a move() command; I wouldn’t expect less out of PHP.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* Tuesday, December 21, 2010 2:20 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] What is equivalent function to mv command in
php for Asterisk Spool directory usage?
Hi Everyone,
I understand that there are a few warnings about using cp to move .call
file...
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Re: [asterisk-users] Recommendation for a Linux based SCADA
Thanks for this info. It seems like good hardware and software solution provider. I'll explore it a bit more and see if it fits my client's need. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-12-20 9:56 AM, marvin horst fivehor...@gmail.com wrote: I'm not certain what you mean by needing to setup up a SCADA solution? I assume you want to connect an industrial data acquisition and control system to Asterisk. We have a SCADA system interfaced with Asterisk in our facility. The SCADA hardware we use is the SNAP PAC system from Opto22http://www.opto22.comwhich provides a linux SDK http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 . You also can set the Opto hardware to send SNMP messages on certain conditions. On Wed, Dec 15, 2010 at 4:23 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous e... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marvin Horst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendation for a Linux based SCADA
Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone
DTMF sent from cell phones are usually not well recognized at the asterisk end. The main reason for this is that cell phones transmit out-of-band DTMF, which by the time reaches an asterisk server traveling through cell towers, their equipment, various VoIP carriers etc. is usually drifted away from its acceptable frequency threshhold. Or if a carrier is converting it into inband, it might not be right at this carrier's end, meaningful it'll have no tone at all. Receiving out-of-band DTMF over physical lines like T1s is usually much more reliable than SIP, because the expensive equipment at big telcos is better at fixing up bad tones and send you the correct tones. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-05 11:24 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *John Regal *Sent:* Friday, November 05, 2010 10:11 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Elementary question - accessing feature codes fromcell phone Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or ... Hope this answer is more helpful than harmful, but in my experience and reading, feature codes and cell phones don’t play well together. DTMF processing is usually way less than 100% reliable in this setup. Your best bet is probably to replicate the feature function you want into an extension (1234 instead of *72) and dialing that from your cell or using the web interface on your cell to do the ARI function. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine channels in use from CLI
How about 'show channels'. As for filtering, you'll have to do it separately using a format like: asterisk -rx 'show channels' | grep 'your filter' You can filter the output further using awk. But each filtering will take a second or two based on what you are filtering. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-04 8:35 PM, Michelle Dupuis mdup...@ocg.ca wrote: Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I SIP SHOW CHANNELS I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
If 1.2 is working fine without any problem then why do you need to upgrade to any newer version? I would suggest don't do it. If you really want to do it just for the sake of doing it, upgrade to 1.4 only, which is the most stable and well tested version of asterisk. Upgrading always causes hickups in the new system, and effects quality of service to the customers. As they say, if its not broken, don't fix it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-03 11:30 AM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: satish patel wrote: We are running asterisk 1.2.x version in production environment since ... 1.8 will introduce many features and is the supported standard, which will be important to you... This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite... -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
Its good to know the MATH function because it can do much more and also deal
with floating point numbers. However in your case a simple addition would be
suffice as other posters posted, or try Danny's GotoIf if it fits your
scenario.
Set(vgLabel=vg${MATH(${vg}+1,i)})
Zeeshan A Zakaria
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On 2010-11-03 9:39 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgL...
Use the MATH function.
Philipp
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Re: [asterisk-users] Under heavy attack
Unsuccessful attempts are recorded, however SIP-s is not easily doable on asteridk 1.4. I tried once without any success. Maybe somebody who has successfully implemented it can write a little how-to on it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 4:48 AM, Hans Witvliet h...@a-domani.nl wrote: On Sun, 2010-10-31 at 11:39 -0600, Joel Maslak wrote: To guess an 8 character (which is short) pas... Perhaps this is good enough reason for starting to use SIP-s (using TLS) with large = 2K) keys. Should be safe enough, i think. Snom seems to be capable of handling it, so can asterisk 1.6.x Any unsuccesfull register attempt should add the offending address to your own blacklist (for iptables) hw -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
Its going on and on and on. Nothing like this has happened before. I have several hundreds by now. Make me wish Internet was a more regulated place. Its a place where bad people have the upper hand and good people cannot do anything about it. I know incidences where spammers and attackers were tried to be punished by genuine companies by doing DoS attacks on their zombie machines and as a result these companies got so much DoS that they were left with no choice other than to close their genuine and legal businesses. And when even reputable companies like Amazon become part of this criminal activity, and refuse to do anything against it, what can rest of us do? Nothing, but suffer. Unless main Internet routers will identify these attackers and block their IPs, there is no real way to control this criminal activity. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
And obviously these attackers read our emails on lists like this and adjust their sick strategies accordingly. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
Hi Cary, Can you email me off the list to point it out? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 1:37 PM, Cary Fitch ca...@usawide.net wrote: I was going to point out a failing of the attackers, but figured they read the list and don’t need any more tips. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Monday, November 01, 2010 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FW: Under heavy attack And obviously these attackers read our emails on lists like this and adjust their sick strategi... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
Too late, now switching to attack level: lethal :) No, I am not one of these losers, and don't ever plan to be. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 1:49 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 1 Nov 2010, Zeeshan Zakaria wrote: Hi Cary, Can you email me off the list to poin... Don't do it! Zeeshan might be an attacker!! :) Just kidding Zeeshan. Couldn't resist. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
Finding and punishing the abusers is the real problem, specially when in my country (Canada) where we generally don't like punishing people (or they get away finding loop holes in the law, or thanks to their lawyers), how would we catch people in other parts of the world and punish them? Apparently wilderness of the Internet is protected by law and law makers everywhere want to keep it this way. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 1:56 PM, jon pounder j...@inline.net wrote: On 11/01/2010 01:44 PM, Nyamul Hassan wrote: I think the only real solution here is to make people take more responsibility for their actions - find and punish the actual abusers - make users liable for damages caused by infected PC's - defaults from an isp should be everything locked down but with user able to request more ports being opened at no extra cost, if a user asks for it they then take on responsibility for the use of that port. LOL On Mon, Nov 1, 2010 at 23:33, Cary Fitch ca...@usawide.net wrote: I was goin... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Under heavy attack
My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
My count has reached 100 for the day. The server serves doesn't serve international calls anyways, I wonder how would it benefit any hacker in any way. -- Zeeshan Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote: No. It seems that opening up some sort of automatic blocking could cause an attacker forging packets to block legitimate endpoints. It also seems like they won't get in with good passwords, so it isn't actually accomplishing something to worry about the script kiddies if you have good passwords. And this blocking won't actually stop someone with a zero day attack or who is sophisticated and can attack from many IP addresses - these are the real threats for people with good passwords. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF in Asterisk 1.4.*
Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-29 5:15 AM, Asterisk User an.asterisk.u...@gmail.com wrote: Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] being bombarded with SIP packets
Two incidents in two weeks is not bad. I get 2-4 a day. There must be many here with even more than that. You should start considering some safety practices like disabling long distance and international calls by default, put a cap on long distance and international calls even for genuine users, and who don't want to have caps, get their consent that they'll not argue with you if their accounts are hacked. Probably do prepaid billing at least for long distance and international calls. Other than that, fail2ban is a must have. Detailed installation instructions you can find at voip-info.org website and also in my blogs at ilovetovoip.com. Regards, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-28 3:48 AM, Per Jessen p...@computer.org wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No media being sent in SIP call
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also pasting your sip.conf here would be helpful. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-27 6:16 PM, Mike Diehl mdi...@diehlnet.com wrote: There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote: Hi! I've turned off t Take care and have fun, Mike Diehl. -- ___... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan help
Chapters 4, 5 and 6 is a good start. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:aster... -- _ -- Bandwidth and Colo... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
Actually it is bad only when received on cell phones. Today I listened to the same voices on a Cisco 7942 and they were great. I actually enjoyed listening to them. Not bad on X-Lite either. Previously I was mostly listening to them only through cell phones. So it means it is because of the transcodings at cell phone providers' ends. Bad though because many customers use cell phones exclusively. Maybe if I convert them to gsm format before playing, they'll play better, but will add delay and additional processing because they are converted and played in real time. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and ... -- _ -- Bandwidth and Colocation P... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan help
I totally agree with Steve's wise advice. One should at least give himself a week learning asterisk fundamentals and related Linux basics before jumping into creating dialplans or setting up Telecom systems. Asterisk's official book's first few chapters cover all the basics which every asterisk user must to know. Otherwise seeking help here won't help because you won't be able to even understand the answers here. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-24 7:59 AM, Rayan Smith rayan.o.sm...@gmail.com wrote: Hi Jigar I am facing issue while generating a dial plan for the following case: all caller should be as... Try DISA component, and then use MeetMe component if you want callers to go to conference or Dial component if you want them to go to extension. I have created a dial plan using vdp I tried submitting it here but I don't know how to extract t... Visual dialplan outputs standard extensions.conf code. You can get the code by selecting Local deploy option at preferences window or SSH to Asterisk server and check extensions.conf. I was coding dial plans in vi for some time and then switch to Visual Dialplan, much easier and faster, very useful tool. Rayan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 and T1 on the same card, or on the same server
Thanks Kevin to verify this. This would really solve a very big problem for me as E1-T1 conversions has been a big part of my work lately, with no satisfactory and reliable solution yet. I'll propose this card to my client and would love to try it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-22 6:15 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 10/22/2010 04:05 PM, Zeeshan Zakaria wrote: Hello list, (Resending this email due to a typ... Yes, the cards in question can handle some ports configured as T1 while others are configured as E1. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality
Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral voice quality not good
Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Licensing of Default MOH
I think you are the first person ever to ask this question. Of course you can use them, they are royalty free for a purpose. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-22 5:53 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Licensing of Default MOH
I didn't know about Digium's cool case studies. Will my realtime virtual PBX with partially javascript based GUI and Voice Reminder service fit into cool case study? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-22 7:17 AM, Andrew Latham lath...@gmail.com wrote: The sound files for MOH, just like the voice files of Alison and others are open and free. You of course can always donate your royalty free sounds or pay for some new sounds. If your language is not included in Asterisk, please contact a quality voice actor and submit some sound files for your language. If you are using Asterisk and the sound files in some unique or large installation you may want to send a summary of the project to Digium so that they can add it to the list of cool case studies. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Oct 22, 2010 at 6:44 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: Hi, I wonde... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Licensing of Default MOH
Thanks for this info. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-22 7:45 AM, Andrew Latham lath...@gmail.com wrote: Have a look... http://www.digium.com/en/company/casestudies/ Contact John Todd jt...@digium.com with your case studies... ~ Andrew lathama Latham lath...@gmail.com On Fri, Oct 22, 2010 at 8:26 AM, Zeeshan Zakaria zisha...@gmail.com wrote: I didn't know about D... _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same extension registering over eth0 and eth1
Rob, you are the man. Thanks for pointing me in the right direction. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-22 12:28 PM, Rob Coward r...@jive-videos.net wrote: Any reason you cant change the asterisk server to bond the 2 nics together ? We use bonded nics a lot to provide resilient networks, and as far as any apps on the server are concerned, you are only talking to a single interface bond0 instead of eth0 and eth1. Rob On Mon, 18 Oct 2010 17:03:45 -0400, Zeeshan Zakaria zisha...@gmail.com wrote: I didn't desig... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 and Pt on the same card, on in the same asterisk box
Hello list, I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran) Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 and T1 on the same card, or on the same server
Hello list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran, dealt with them in the past, won't deal again.) I talked to Digium and the sales guy said their TE420 card can *supposedly* do it as it can have ports configured as a mix of E1s and T1s. Has anybody used this card for the purpose of T1 to E1 conversion? Regards Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a new server
I think I'll prefer Dell over supermicro, as another customer I worked for always complained about supermicro. I also once used supermicro and I had no luck with it. But which model of Dell is good for this requirement? I don't want to get over powerful server than required for this setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 6:56 AM, Andrew Latham lath...@gmail.com wrote: No transcoding? OK, this will work... http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, W... -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS always returns NOANSWER
Maybe you should post this portion for your dialplan. I have done the same thing several times and never had this timeout issue. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 4:08 AM, GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com wrote: Hi, Here is the scenario: 1. 1st phone calls and asterisk dials to extension no. 2. Extension answers 1st caller(which makes it busy). 2. 2nd phone calls and asterisk dials to extension no. 3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in DIAL cmd) before proceeding to the next step in dialplan 4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY the problem is, since the 2nd caller hears a busy tone, it should not wait for the timeout to expire, and proceed immediately in fetching the DIALSTATUS. I also tried this scenario and used DEV_STATE, but it always returns NOT_INUSE I already assigned qualify=yes in my sip configuration but still to no avail. any ideas? regards, RYAN ICASIANO -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
I was thinking on the same lines, i.e. setup a server which will be regularly updated with these bad IP addresses, and anybody looking to block bad IPs will be able to get this list from here. For example when I get mail from Fail2Ban (which I am getting more and more everyday now), a copy would be sent to this server with the updated bad IP address. But the problem is how to make sure that only legitimate users are contributing to this list. Contributors to this list somehow need to verify to an admin that they are not hackers, and this the hard part. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 11:46 AM, Steve Howes steve-li...@geekinter.net wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendation for a new server
Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a new server
Any suggestions? On Wed, Oct 20, 2010 at 9:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clustering
How about setting up a high availability cluster using DRBD and Heartbeat? There is some good info on it on the Internet. In this type of setup you have two exact same servers running in parallel, and only one has the required services up. They keep themselves in sync. When the primary one goes down, the secondary instantly takes over. Active calls are though dropped, but after that everything is back to normal. There are various other options regarding which server will stay primary, or how and which services will be used on which server. Another option I am exploring is using the same thing but in Proxmox with DRBD. Somebody told me it could be setup so that even the active calls are not dropped. I haven't set it up yet, but will try it when get time. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-18 10:59 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham Sent: Monday, October 18, 2010 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] clustering Unfortunately we are too late to switch to Kamailio. I mean we have developed our pbx with call features and routing on asterisk only. If we switch to some other software that means we will have to redo a lot of development again. I was thinking of using DUNDi and distributing the registrations on different servers. I just dont get one point. lets say if i have 2 users registered on different asterisk servers and... snip Sorry for second post, but I have a Polycom 501 registered to 3 servers. I hit the line button and if the server I pick is down, I don’t get a dial tone. Hope this is useful. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clustering
Hi, I have worked with only two servers setup, don't know how it would work in three server setup. You'll need to do an experiment, but know that it won't work if you have T1 lines. HA and DRBD is good for pure VoIP. Before the end of this year hopefully I'll be setting up two more redundancy solutions, and will try some new techniques, and probably try three server setup too. At that time I plan to post a tutorial on redundancy solution on my blog, because seems like a lot of people want to know how to do it, yet guidance is very limited. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-18 12:49 PM, Rizwan Hisham rizwanhas...@gmail.com wrote: Hello Zeeshan, How about doing the mixture of what I want to do with your strategy. I mean, what if we have 3 asterisk servers with distributed registrations and also have heartbeat installed monitoring all the servers? will that work? On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria zisha...@gmail.com wrote: How about setting... -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Same extension registering over eth0 and eth1
Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only from eth0, and if this port fails, sends registration coming in from eth1? Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same extension registering over eth0 and eth1
Will OpenSIPs do the job? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-18 4:43 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Oct 18, 2010 at 3:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Is there a way/softwa... DNS SRV or a SIP proxy. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same extension registering over eth0 and eth1
I didn't design the network, it was already here at clien't site. It is designed for redundancy. I am trying to come up with a solution to make asterisk work in it. I am looking into opensips how it can help me. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-18 5:00 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Oct 18, 2010 at 4:47 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Will OpenSIPs do the ... Any proxy would work, however I would re think your network design. Re-registering the same phone, with the same extension, on the same PBX is asking for trouble. If you want to do redundancy, I would set your network so only one ethernet route is active at one time, then it is a matter or routing. If you want both ethernet ports active, then you are doing load balancing. Something Asterisk by itself is not strong at. Hence the SIP proxy or DNS SRV records. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-17 3:37 AM, Dan Journo d...@keshercommunications.com wrote: Nope, Its a totally normal self-built Asterisk. Dan Zeeshan Zakaria zisha...@gmail.com wrote: Do you use FreePBX by any chance? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-16 6:38 ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Do you use FreePBX by any chance? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-16 6:38 PM, Dan Journo d...@keshercommunications.com wrote: Serious answer: Looks like a process running asterisk -r. Do you have any sort of AGI, cron j... Thanks for lightning my day! Is there any way to debug this because as far as i'm aware, there's nothing running that command, (except for me) -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
I never had this problem, and this is certainly not asterisk's fault. Probably your conversion is not good. Can you email me a file and I'll do conversion on my end, and if sounds good, let you know how I did it. Then a script can be written to convert them all. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:25 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote: I am about to have to dump Asterisk in f... Can you post a link to a sample before and after file as well as the command line you are using to convert the file? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fraud advice
For future I would highly recommend to have at least fail2ban installed. This way sipvicous IPs will be blocked instantly before they could create any damage. Also I prefer to limit International calling to only certain limit, e.g. only for $10 per account, but this depends upon how your business deals with international calls. I get a few IPs blocked everyday by fail2ban, though by default no new connections are allowed international calls on my system. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:40 AM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to n... Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime users call problem
Check sip_buddies table for the correct context entry. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 5:31 AM, Oguzhan Kayhan oguzh...@bilkent.edu.tr wrote: Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not available voicemail activates etc. But when i call a realtime user which is already on peer list i got chan_sip.c:20152 handle_request_invite: Call from '' to extension '' rejected because extension not found in context 'DLPN_WorldcallDial'. And this is when i call a static user (works normal) Executing [6...@dlpn_worldcalldial:1] Macro(SIP/-001e, stdexten,6000,SIP/6000) in new stack This is dlpn_worldcalldial [DLPN_WorldcallDial] include = default include = CallingRule_worldcall include = parkedcalls include = conferences include = ringgroups include = voicemenus include = queues include = voicemailgroups include = directory include = pagegroups include = page_an_extension Thanks a lot if you can tell me what to check -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protocol or using DAHDI/Zaptel. Could you explain a bit what type of setup you have? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 9:15 AM, Dan Journo d...@keshercommunications.com wrote: Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding? Once you are sure that asterisk is receiving DTMF fine, then you should ask your provider what DTMF setting you should have on your system. Usually all of them support RFC2833, so if in your sip.conf where you have defined the trunk, dtmfmode is set to rfc2833, your provider should receive it and pass on to the next carrier or trunk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 10:19 AM, Dan Journo d...@keshercommunications.com wrote: It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protoc... Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] checking CDR
Hi, (Following is for asterisk 1.4) For the forwarded calls, you should see two entries in the cdr, and this is because a forwarded call is actually two separate calls. You have to look in the channel and dstchannel fields of the cdr to match the call ids of the calls to figure out which calls were forwarded. Incoming call's channel value and outgoing call's dstchannel value will be the same, except a comma and digit at the end, showing if it was the first call on that id, second, third or more. I have programmed two billing systems, and this is how I catch forwarded calls and bill them, works perfectly fine. Though it is confusing. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 1:21 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Action Originate
You need to create a dialplan context to achieve it and then access it using originate. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-11 5:54 AM, 施铁泉 justhin...@gmail.com wrote: I use the action Originate，i want the called first ringing，the called answer,callee ringing.it can achieve? Best regards, justhinker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
For a production environment, 1.4 is the most stable, and it has everything one needs to setup a telecom platform. As per my understanding 1.6 never got the same recognition for stability as 1.4, plus it doesn't have any significant advantages over 1.4. The newer version 1.8 series might be my next jump once it'll be out of beta, but at this time it should not be used in a production environment. Many of us still use 1.4 in production and if you are just starting, this'll be your best choice. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Be... *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham *Sent:* Wednesday, October 06, 2010 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Difference Is there any major architectural difference between 1.4 and 1.8? The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from , to |) and the AGI structure is enhanced. If you don’t use AGI’s, a qualified “not really”. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference
Here is a presentation from Kevin P. Fleming, Director of Software Technologies at Digium. Information might be old by now still gives a good overview of what is new in 1.6: http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf Summary of his presentation is as follows: – Asterisk 1.6 contains much new functionality, although nothing revolutionary – Asterisk 1.6's core has been improved in many ways that will reduce the performance impact of new features being added and also the likelihood of difficult to find locking and data structure bugs – Future releases of Asterisk 1.6 (1.6.1, 1.6.2, etc.) will get new functionality as well, in a controlled fashion – Asterisk 1.6.0 is not recommended for production usage yet, but we would very much like users to try it, report problems and help test the product in more scenarios than the development can test themselves -- Zeeshan On Wed, Oct 6, 2010 at 12:12 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: Back in the days i heard that they have changed the architecture in 1.6 and its a lot better than 1.4 (6 times better call handling and robust architecture, someone told me). If they have decided to take the 1.6 architecture to the next level in the new 1.8 version then its a good thing. On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 6 Oct 2010, Rizwan Hisham wrote: Is there any major architectural difference between 1.4 and 1.8? Nope. The developer's just got tired of typing .4 Of course, the joke's on them -- 1.8 is only .4 better than 1.4. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Qureshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to learn encrypted VoIP development for embedded systems
Hi list, A few times I have been asked if I could do encrypted VoIP development, for embedded systems, and in C++. And my answer has been in negative. Now I am thinking I should start learning how to do it, but I have no clue where to start from. I have been developing in Java for some time now, but haven't touched C++ in years. I haven't programmed for embedded systems. Even if I knew C++ well enough, I have no idea how to program my own protocols and then also come up with some encryption methods for them. I'll appreciate if those of you who have experience in this field could guide me to any references, links, books, or other learning sources. Sincerely, -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
You can use proxmox from proxmox.com. I am using it for the same reason you want to use it. I have been testing it for some time now and it works great. Proxmox is an excellent hypervisor and it is free. Easy to install and simple to setup. Install it drom its ISO. Then you can download a OenVZ CentOS 5.2 instance for it from proxmox website, install it, give it an IP address and you have your server ready. Install on it asterisk as you would on any other system. I have detailed instructions for it on my blog, which I documented when I was setting up asterisk from scratch on a CentOS instance on proxmox. Once you have asterisk all setup, you can simply copy/paste the folder with virtual machine instance using a new name, and you have a second copy of your asterisk setup. Assign it a different IP address. I created 7 copies of my main setup, each with its own IP address. Proxmox also gives you option for hardware level virtulization, called KVM. I haven't tried it. With only OpenVZ you shall not be able to use zaptel/dahdi hardware though, and I don't know if KVM allows for it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-05 10:57 AM, Steve Howes steve-li...@geekinter.net wrote: On 5 Oct 2010, at 15:13, Gordon Henderson wrote: $ /home/asterisk1/usr/sbin/asterisk -g for firs... More than one IP on the box. Change the bind address.. Easy, no? Steve -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempts to hack Asterisk - What do these lines means
Seems like anonymous SIP calls which end up in from-sip-external context with a dead end. This is usually how hackers start their hack attempts. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-02 3:05 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker is trying to reach: 02/10/2010 11:10 SIP/113.105.152.51-00fb sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fe sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fc sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fd sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00ff sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-0100 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0101 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0102 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0103 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0104 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0105 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0106 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0107 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0108 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0109 sip sip sip s ANSWERED 13 Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] should trixbox system hang when ISP drops connection?
Is your ISP doing DNS resolutions for you? If yes, then I also think it has something to do with the DNS queries which hangs asterisk. But it should not bring the server down. On CentOS, caching name server should be very easy to install by doing: yum install caching-nameserver I don't remember if it also sets up the required config files. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 11:15 AM, Warren Selby wcse...@selbytech.com wrote: On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day rpj...@crashcourse.ca wrote: so, is there... Try installing a local caching nameserver on the same box that runs asterisk, and have that handle DNS queries for you. I remember at one point that trixbox would hang if you had any SIP trunks configured and you lost internet connectivity, but a caching nameserver on the same box tended to help. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best format for playback/generation
If your sip provider supports gsm, then it is fine to send them your existing format, but I am sure by the time voice reaches an end user, it is transcoded at least once or twice again, so you can never guarantee what quality the end user is getting. I would stay with ulaw, as it has more chances to retain a better quailty even after a few transcodings, plus almost every sip provider will be able to receive it as it is and pass it on as received. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:02 PM, Gareth Blades list-aster...@skycomuk.com wrote: The best format would be in whatever format asterisk is sending the final audio out in. Even if you store it in the highest quality asterisk may have to transcode it on the fly so its best to store it in an already transcoded format to reduce the cpu load. For dahdi you would want to use the native .sln format. For sip use whatever coded you use over the sip connection. Danny Nicholas wrote: Greetings fellow listers, I have an ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
Its a long and old thread, haven't read it all, but just to let you know this happens when there is no reply from the DNS. So change DNS or install it locally on your asterisk server. At least caching name server should be installed. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:51 PM, Gopalakrishnan A.N sai...@gmail.com wrote: Still I have the connection loss when internet goes down, I have to restart the Asterisk machine or need to remove the VoIP trunk accessing internet... DNSmasq is the only option by losing the connection when internet goes down...is there any other way... Thanks On Fri, Feb 12, 2010 at 4:20 AM, Matt Riddell li...@venturevoip.com wrote: On 9/02/10 12:59 ... -- Thank you with regards, Gopalakrishnan A.N, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realm: security issue
From what you explained it seems to me that your mobile provider has blocked your sip communication altogether. Have you tried changing IP address of your asterisk server? If changing IP works, then probably your provider has blocked you sip communication by IP only. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-23 7:22 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I have my friend that use his mobile (Nimbuz) to connect for the Asterisk and his account was working fine. Suddenly it stop working (not able to register). From my mobile (Nokia) I was able to register using my username and password, so I tried to register using his (my friend) username and password (that was using them from Nimbuz), it did not work. I come back trying to register using my origin username and password (which was working fine just before a while), it did not work. I removed my username and my friend username from the Asterisk and then I created a new username and password (different than all other) and I tried to register from my mobile, also it did not work !!! I start beleive that it is something related to detecting a hacking (maybe Nimbuz does not use a good security), this caused the MAC to be considered as hacked. Please, can someone advise me how to resolve this problem? Where I can find those MACs that need to be removed from block list? What can I do to get out from this problem? Any advise? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
Have you tried removing option 'g' from your Dial command? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote: Hi, I use asterisk with sip3000 device with sip-aho connected to PSTN and sip-ahi connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten = 99,n,Dial(SIP/sip-ahi,30,g) exten = 99,n,Hangup() The asterisk properly detects hangup of the caller as I see following lines in asterisk -crvv ... Dial(SIP/sip-aho-0003, SIP/sip-ahi,30,g) == Spawn extension (from-pstn, 99, 8) exited non-zero on 'SIP/sip-aho-0003' ... How can I make the phone stop ringing the moment caller hangup? -- Arie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension continues ringing after caller hanged up
Do you mean spa3000 or sip3000? I remember having same problem with spa3000 and the problem was somewhere in the settings of spa3000 that wouldn't stop ringing the phone. I don't remember the details at this moment as it was long time ago, but this much I can tell that it is a config issue with spa3000 device, not asterisk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 11:02 AM, Arie Skliarouk sklia...@gmail.com wrote: Hi, On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote: Have you tried removi... Of course, with the same result. -- Arie Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-20 7:45 AM, Arie Skliarouk s... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration attempts
It means that fail2ban is not configured correctly on your machine. For me it works fine, and in fact lately these registration/hack attempts have gone up significantly, thanks to cloud computing I guess. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-17 5:28 PM, dave george dgeo...@teletoneinc.com wrote: I am getting several hundred registration attempts on my aserterisk per minute. I have fail2ban installed but it's not stopping the attempts. Any suggestions. Whatever they are using is changing the userid on each attempt. Latest IP: 209.172.57.219 Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
When making an outbound call, if sip peer is not registered, first it registers itself, and then makes the call. This is why you don't see any problem dialing out. For receiving, asterisk has to wait until the sip peer registers, otherwise asterisk has nowhere to send the call. I know the pain, as I deal with the same situation. So I don't do 'reload' or 'sip reload' except if sip password (secret) has been changed, in which case I prefer to use 'sip prune realtime peer extension' followed by 'sip show peer extension load'. Most of the sip devices re-register every 60 seconds, or if they don't on a realtime network, depending upon the bandwidth, they should be made to do so. Or in some cases you can send a reboot signal to a sip device too. The bottom line is, try not to do a 'reload' as it would affect everybody else too by dropping their registrations temporarily. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-16 10:04 AM, Peder pe...@networkoblivion.com wrote: A reload flushes the SIP registration database, so once you do a reload, that phones reg is gone. If the reg is set for a short period, say 60 seconds, then in 60 seconds it will re-register and work fine. Yes, it is a total pain, but this is the way it has worked since day 1 for realtime. I agree that it seems wrong and even argued that several years ago when this feature came out, but it is what it is. As someone else said, the answer is don't do a 'reload', do an extensions reload or whatever it is specific to your changes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-bo... Sent: Thursday, September 16, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussi... Subject: Re: [asterisk-users] Bug with Realtime? That's not a bug. Only when the phone registers ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Purpose of qualify=yes
I prefer to keep qualify=on for all the extensions, as it gives you an idea which extensions are going to give you trouble. For extensions with qualify value greater than 300 ms you should definitely worry. For extensions at 2000ms delay or more, turning qualify off simply means to ignore the obvious problem. Such extensions have communication or network issues which require serious attention. You can set this parameter to, e.g. 3000 ms or more if dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall even at 2000 ms conversation is not truly real time and not easy. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-16 11:38 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Chris Owen ow...@hubris.net writes: So I guess my question is what is the real purpose of the q... The purpose is simply to see if the phone is available. For your particular use it is likely best to simply turn it off completely. If a phone disappears, its registration will eventually time out anyway. /Benny -- _ -- Bandwidth and Colocat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi, I went over your dialplan and though it looks fine at first glance, but because I have no experience with Asterisk 1.6, so I would like to ask if commas in mysql query are ok without escape character? In my asterisk 1.4 I would type it like: SELECT var1\, var2\, var3 FROM ... Other things which come to mind: 1. Is your MySQL up to date? 2. Software versions on your test system are the same as on the production system? 3. Can you post a MySQL query from your dialplan which works fine. Regards, Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 9:20 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 09/15/2010 02:45 PM, Steve Howes wrote: On 15 Sep 2010, at 13:22, Jonas Kellens wrote: ... Off course I did that, Steve, before I did a locate on 'core'. But doesn't locate also have some PATH ? Where in my case /tmp is not in it. Meanwhile I have come across this : 1. start Asterisk with safe_asterisk 2. enter gdb asterisk core. 3. enter bt while in gdb (or do a bt full) 4. enter thread apply all bt I have no experience with this, so I post my output : [r...@asterisk ~]# gdb asterisk core.4483 snip This GDB was configured as i386-redhat-linux-gnu... /root/core.4483 is not a core dump: File format not recognized (gdb) bt No stack. (gdb) Quit Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
You can do 'extensions reload' or 'ael reload' if you don't want to lose real-time sip registrations. I only reload what is needed to be reloaded instead of reloading everything. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 4:28 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-09-15 03:41 PM, Dan Journo wrote: I think ive found a bug but need someone to double check. ... That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
In theory it should work but in real life it doesn't. Converting reliably half an hour of speech into text is simply a dream. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 4:52 AM, Nickolay V. Shmyrev nshmy...@nexiwave.com wrote: В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA пишет: Thanks for update. is there any command for using sphinix to convert speech to text Yes, first of all make sure you compiled latest snapshot. Then run # sphinx_lm_sort lm_giga_20k_nvp_3gram.arpa lm_giga_20k_nvp_3gram.arpa.sorted # sphinx_lm_convert -i lm_giga_20k_nvp_3gram.arpa.sorted -o lm_giga_20k_nvp_3gram.lm.DMP This will create a language model lm_giga_20k_nvp_3gram.lm.DMP And finally convert audio pocketsphinx_continuous -infile your_audio_file.wav -samprate 8000 \ -hmm Communicator_semi_40.cd_semi_6000 -lm lm_giga_20k_nvp_3gram.lm.DMP -- Nexiwave - Speech Mining Solution For Call Centers http://nexiwave.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
If it works even half decent, kindly post your result on the list. I need something similar for a client, that is voicemail to text. After my research my proposal was to hire someone to listen to voicemails, type them and email them, as I couldn't see any way to do it, though in theory any good voice recognition engine should do it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 7:09 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: is it possible with lumenvox i will purchase liceance regards Dhaval On Tue, Sep 14, 2010 at 4:20 PM, Zeeshan Zakaria zisha...@gmail.com wrote: In theory it shou... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Random File Name
I use ${UNIQUEID} with ${CDR(accountcode)} and it works great. But this is
when you have an accountcode for each user. As the last poster suggest, you
can append it with date and time and it'll be truly unique and also help you
keep track of the recording.
Zeeshan A Zakaria
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On 2010-09-14 9:54 AM, Dan Journo d...@keshercommunications.com wrote:
Hi,
Im looking at using MixMonitor to record calls and I know that I need to set
the filename first.
However, with the number of calls coming in, hard coding the filename isnt
an option.
So I need to do something like this:-
MixMonitor(RANDOMNUMBER.wav)
But can't find a way to generate a random number.
I thought that maybe I could use a unique variable that already exists for
the current call, but I'm not sure which one to pick.
Can anyone give me some advice on this?
Thanks
Dan
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Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code. http://www.geekzone.co.nz/sbiddle/7183 http://www.geekzone.co.nz/sbiddle/7183-- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. It's very simple to find the actually issue, if the OP does the following: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Random File Name
Can you post what are you doing to see UNIQUEID? And also what version of
Asterisk you are using?
Zeeshan A Zakaria
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On 2010-09-14 12:11 PM, Dan Journo d...@keshercommunications.com wrote:
${UNIQUEID} is going to be realtivly unique certnely in the short term
I dont understand something. When I do ${UNIQUEID}, I get something like
this:-
SIP/215.166.5.140-0bbf
Is this correct? Its not a valid file name.
Thanks
Dan
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Re: [asterisk-users] Random File Name
Or
1234 = {
Verbose (ID is ${UNIQUEID});
};
:)
Zeeshan A Zakaria
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On 2010-09-14 12:27 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Tuesday, September 14, 2010 11:15 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Random File Name
Can you post what are you doing to see UNIQUEID? And also what version of
Asterisk you are usi...
On 2010-09-14 12:11 PM, Dan Journo d...@keshercommunications.com
wrote:
${UNIQUEID...
My guess is that “Dan” is doing something like this
Exten = 1234,1,Verbose(ID is ${EXTEN})
“SIP/215…” is an ${EXTEN} value
Should be doing something like
Exten = 1234,1,Verbose(ID is ${UNIQUEID})
This should return something like
120003949488.0003
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Re: [asterisk-users] sip show channels
Do a 'show channels'. You can also do 'show channels concise' or 'show channels verbose' for more details. In any case, it'll show you number of active calls at the end of output. Now some may point out to prepend 'core' before issuing these commands. I prefer to be brief, and used to this shorter syntax. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 12:39 PM, Dan Journo d...@keshercommunications.com wrote: Hi, I'm trying to view a list of the active calls to see if I can restart Asterisk. When I do 'sip show channels', I get a huge list like this (just a sample pasted):- 92.110.7.210 (None) 198827f2469 00102/0 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 6b211bb04ac 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 4e05e84848b 00102/0 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 7bdb88176f0 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 49ef531c4b7 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 4039350f335 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 0b91ed9d733 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 3223de36008 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 258c01bf4d6 00102/0 0x0 (nothing) No Init: OPTIONS 92.108.34.153(None) 2f35c6eb767 00102/0 0x0 (nothing) No Init: OPTIONS Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show channels
True, that is even better. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 12:49 PM, Steve Howes steve-li...@geekinter.net wrote: On 14 Sep 2010, at 17:32, Dan Journo wrote: I'm trying to view a list of the active calls to see i... Don't?. 'core restart when convenient' will wait until there are no calls. S -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How different is implementing Cisco based system than Asterisk based system?
Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me to implement for his project a Cisco based VoIP system. I told him that I specialize in Asterisk based systems, but he is not even aware of Asterisk. The requirement of project is such that chances are slim that this firm will consider Asterisk based system. So I told him that though not experienced specifically in Cisco, if they hire me I'll setup their VoIP on Cisco system. Now I have no previous experience with Cisco systems and don't want to screw up anything. Are they much different than Asterisk based systems? I guess the underlying VoIP technology is the same for both the systems so it shouldn't be hard to set it up on Cisco. Any ideas, suggestions. I'd appreciate your help as what to look for, where to start from. My experience with Cisco is limited to their networking equipment, IOS, their 7960 series phones and making them work with asterisk, and also using Cisco press's wonderful book 'Taking charge of Your VoIP Project'. Sincerely, Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?
I also thought that they should get it from an official Cisco reseller if they wanted support. Maybe at this stage they themselves don't know what they want. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 4:14 PM, Peder pe...@networkoblivion.com wrote: My best advice would be “don’t do it, it will only cause headaches”. It is completely different than * with different terminology, design considerations, etc. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Tuesday, September 14, 2010 2:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] How different is implementing Cisco based system than Asterisk based system? Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?
I'll keep this all in mind. I don't plan to become a Cisco expert over night. Flirts I'll try to make them use Asterisk. I don't know the details yet. But some of these big organizations don't even want to consider anything other than the proprietary systems. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 4:23 PM, David Backeberg dbackeb...@gmail.com wrote: On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Now I have no previous experience with Cisco systems and don't want to screw up anything. Are th... sometimes. Cisco supports SIP, but depending on the product, asterisk inter-networking with call transfers / dials / etc. can be, ummm, interesting, and you have to do Transfer() rather than Dial() if you want subsequent transfers / conferencing, etc. to work within Cisco. Basically, call setup / control and RTP aren't necessarily on the same device(s) which is the opposite of my asterisk experience. I guess the underlying VoIP technology is the same for both the systems so it shouldn't be hard... sometimes. Again, the size of the deployment is relevant here. Any ideas, suggestions. I'd appreciate your help as what to look for, where to start from. Again, at some point you'll need to call a reseller. In the meantime, if you want to keep them honest, you should get your hands on paper or digital copies of the Cisco press books about their phone system products. Can't recommend anything specific without knowing things about size and purpose of the install. -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
It is simply not possible, though it might be in the distant future. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 1:50 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Thanks Paul, i think still i have some problem to understand , i mean to say that i have 30 minutes audio file in WAV format and i wnat its text here are the scenario . - Call comes in - start recording - call remains for 30 minutes - stop recording - convert wav file audio to text. is this possible with lumenvox or any other engine. regards Dhaval On Tue, Sep 14, 2010 at 10:57 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
This is not elastix or FreePBX forum and asking non-asterisk related questions here is misusing this mailing list. Allow anonymous sip is not an asterisk feature. Look in the code in extensions.conf what it is programmed to do and you'll figure out why it is happening. Or maybe post the code and ask why such a behaviour, which'll be better way to ask this elastix related question here. If you know what this part of dialplan does, rest is easy to figure out. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Mr. John, This is not about policing and this is asterisk-user mailing list. Poster is a FreePBX user. I am very well aware of Asterisk IS involved, but the fact is this is not a FreePBX mailing list. If the poster examines the problem code from extensions.conf, or post it here, it'll made him and everyone clear why is it happening. But poster apparently not well verse in Asterisk anyways. FreePBX has their own forum as well. Or maybe you can explore FreePBX code for him. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 9:40 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce bruceb...@gmail.com wrote: I have a provider whose... Have you considered contacting your provider? I would think that is your first step. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
I think this may be because ... So you think, don't know. Maybe you knew if you knew the FreePBX code, or bothered to look into it. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
So you are sure it has NOTHING to do with extensions.conf. This clearly
shows your absolute ignorance about what poster is asking and how FreePBX
works. Had the problem code been posted, this problem would already have
been solved by now.
And sorry if you think this is policing. You can think whatever you like.
--
Zeeshan
On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous sip is
not an asterisk feature. Look in the code in extensions.conf what it
is programmed to do and you'll figure out why it is happening. Or
maybe post the code and ask why such a behaviour, which'll be better
way to ask this elastix related question here. If you know what this
part of dialplan does, rest is easy to figure out.
Zeeshan A Zakaria
Heh - listen to you - top posting, bad english, and self appointed list
police. His problem certainly seemed asterisk related to me, and has
NOTHING to do with code in extensions.conf. He even posted CLI commands
he is attempting to use to find his problem. I applaud him for taking
the initiative to try working it out on his own, and see no problem at
all with his question. I hope we can help him fix it.
j
--
www.ilovetovoip.com
On 2010-09-10 11:17 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working. Nothing has been changed on our end.
Here is what I get when doing sip set debug peer PROVIDER:
Sending to 123.123.123.123 : 5060 (no NAT)
That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly
and you see the ACK, REQUEST and ACCEPT of sip debug just fine.
This is Elastix with Asterisk 1.4.33.1
Any thoughts?
Thanks
--
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Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Actually it is a very easy to understand and fix issue, but looking at the code taking care of anonymous sip calls is the key. Those who post third party GUI related issues should at least post the underlying asterisk config or code here, so the asterisk part of the problem can be fixed. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote: Sending to 123.123.12... Either you changed the peer parameters or they did... If he is not receiving any response, it is most likely a routing issue. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger ... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. When there is enough detail in the post and I am aware of the problem, I always try to help. I don't believe in making guesses. Troubleshooting requires some good detail of the problem. And yes, answering non-asterisk related issues is not the goal of this mailing list. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-11 9:24 PM, Jeff LaCoursiere j...@sunfone.com wrote: -- www.ilovetovoip.com On 2010-09-11 7:22 PM, Paul Belanger paul.belan...@polybea... [un top posting] On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote: Actually it is a very easy to understan... Its not that he isn't receiving a response - its that his peer debug statement isn't getting tripped because the peer hasn't authenticated. That's why I suggested he debug by IP rather than peer. Then what he will see is the SIP auth attempts and asterisk rejecting them, but in my experience not much is of value in seeing those packets - it doesn't point to *why* the connection is being rejected. The routing must be ok since allowing guest sip connections (the result of setting accept anonymous in FreePBX) allows the calls to come in fine. His problem is the peer authenticating. This of course has nothing to do with extensions.conf, as the dialplan is not involved. It is a SIP authentication problem, purely. There is no relevant code to post, and if you had ever looked into FreePBX's relevant code you would realize that it is actually fairly complex, and you would indeed have a difficult time debugging the flow. It *might* help if he posted his peer entry, but without seeing the other side that may not help much either. As Paul suggested first off, he should be in touch with his provider, whose tech support should be able to help him sort it out. I ran into a strange one EXACTLY like this just last week. We have a residential dial-tone customer with a Linksys SPA2102 (our standard device for this service). He had someone come out and replace his home router, and when he did he stopped authenticating. He has a fixed IP, so I enabled the debugging as I have mentioned twice now (by IP) and saw the attempts and rejections. After much hair pulling I *disabled* nat in his peer entry and it suddenly connected fine. This is bizarre, as our standard peer configuration works for 100% of the rest of our customers, who all connect from behind their home nat gateways of all kinds. I still don't know why that fixed it. Sorry you took it so harshly Zeeshan, but the only posts that stick out to me from you are the ones where you are bashing people for posting questions. I don't recall any off the top of my head where you are actually helping. Yup, I consider that policing, and it isn't needed. Like someone else suggested, if you don't want to read it, delete it. And no, I am not going to bother to read back through archives to see if that is the truth. Its my impression of your posts, thats all. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.aste... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openvz
Some days ago in my lab I setup Proxmox, installed a CentOS 5.2 appliance on OpenVZ, installed all asterisk related stuff (except dahdi), including php, mysql, munin, other tools, set it up with a dialplan and it worked just fine. Then manually made multiple copies of the folder where all this installation was stored. This gave me multiple instances of CentOS/asterisk, which I configured with unique IP addresses. I have been using this setup for a few days now and all seems good. I plan to test conferencing next week. So far seems like a good and stable setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-03 1:22 PM, Miguel Molina mmol...@millenium.com.co wrote: Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to ge... El 03/09/10 09:31, mattias escribió: Outlook? Outlook Express is a total PITA. Should I recommend you to use Mozilla Thunderbird... Sorry for the offtopic. And I agree, you should have no problems with asterisk using it inside an openVZ VPS. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL - what is error: ael.flex:647 ael_yylex: Unhandled char(s):
Hi List, When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22 and 1.4.35 respectively, I am getting multiple lines of this strange error: ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s): On three other servers with same versions of asterisk, i.e. 1.4.22, I don't see this error. Number of lines of the error are the same as the number of lines of the code. But the AEL code works fine otherwise. Any idea what is the reason of this error? I couldn't find any useful info on google regarding this error. -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Thank you list for all the valuable input. Based on your input I have decided to stay with 1.4 for now for the production systems. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 2:13 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Gordon Henderson wrote: On Tue, 24 Aug 2010, Roderick A. Anderson wrote: Gordon Henderson wr... Sorry Gordon. No harm was intended. Guess I better stick to lurking. Rod -- Gordon -- _ -- Bandwi... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
That's what I understood too from this one and probably only related google search result, but even if I have just 3-4 lines of code, the error is still there. It is all English characters, so UTF-8 compatibility issue should not be there. I am sure there is some small little config change is required somewhere related to AEL, but where, I don't know. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 11:37 AM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Wednesday, August 25, 2010 10:24 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s): Hi List, When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22 and... Any idea what is the reason of this error? I couldn't find any useful info on google regarding th... According to this https://issues.asterisk.org/view.php?id=14022 1.4.X doesn’t have all of the 8-bit handling it should have (something in your ael script is freaking out UTF-8) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
Actually I have found the problem, and leanred some new stuff along with it. Apparently all Linux files have a mime type information stored in them, which can be checked using command: file -i filename For my extensions.ael, which I copied from a different server, the mime type is 'text/x-c' whereas all the other files have mime type 'text/plain'. Now if I create a new file extemsions.maelstrom on this machine which is by default 'text/plain', there are no errors on doing 'ael reload', however using extensions.maelstrom with mime type 'text/x-c' gives errors, though the code works fine. How it got mime type 'text/x-c' on the other machine, Vim which I use there assigned it this mime type. I'll have to fix it there. Now I am trying to figure out how to convert between mime types. A simpler solution is to just copy text to a new file, but would be nice to do a proper conversion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 12:01 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: -- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On B... *Sent:* Wednesday, August 25, 2010 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s): That's what I understood too from this one and probably only related google search result, but even ... Is there any chance that these files were edited on a Windows machine and then copied back to the Asterisk boxes? That is, are there some nefarious ^m characters hiding in there? Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
This info was useful. So now I have more than a year before I can think about switching to a newer version. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 12:34 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-08-24 09:59 AM, Gareth Blades wrote: Zeeshan Zakaria wrote: If you are planning to move then perhaps look at 1.6 to give you a lo... 1.6.2 doesn't give you any longer of a life cycle (theoretically) than 1.4. See the following link: http://www.asterisk.org/asterisk-versions Leif. -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
It did the trick. Thanks a lot for solving this annoying problem. Yes, it was invisible ^M characters, which I could see in Vim, but not in vi. Now 'ael reload' ouput is how it should be. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 1:29 PM, Rodrigo Lang rodrigoferreiral...@gmail.com wrote: Use the command aptitude install tofrodos to install dos2unix. This command get the file and clear the ^M. Regards, Rodrigo Lang. 2010/8/25 Zeeshan Zakaria zisha...@gmail.com Actually I have found the problem, and leanred some new stuff along with it. Apparently all... -- Rodrigo Lang http://rodrigorecipes.blogspot.com/http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL - what is error: ael.flex:647 ael_yylex:Unhandled char(s):
Thanks Steve for clearing this confusion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-25 3:16 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 25 Aug 2010, Zeeshan Zakaria wrote: Apparently all Linux files have a mime type informati... Linux files are just a byte stream. They do not have mime type information stored in them. file works by examining the first x bytes of the file and comparing this to a magic file of rules to guess file types. For instance, a JPEG file starts with 0xff 0xd8 0xff 0xe1. The -i or --mime command line option causes file to display a mime type instead of a more traditional human readable one. The hexdump command will show you the binary contents of a file in a variety for formats: # Create a file with 2 lines $ printf line 1\nline 2\n foo # Dump the file. Note that 0a (newline, aka line-feed) is the line # ending character. $ hexdump -C foo 6c 69 6e 65 20 31 0a 6c 69 6e 65 20 32 0a|line 1.line 2.| # file says it is just ASCII text, like all text files on Unix should be. $ file foo foo: ASCII text # Convert the file to DOS line endings $ unix2dos foo unix2dos: converting file foo to DOS format ... # Dump the file. Note that the line ending characters are now 0d # (carriage return) and 0a (newline). $ hexdump -C foo 6c 69 6e 65 20 31 0d 0a 6c 69 6e 65 20 32 0d 0a |line 1..line 2..| # file says it has funky line endings. $ file foo foo: ASCII text, with CRLF line terminators CRLF = Carriage Return, Line Feed -- think of a typewriter. Watch the History Channel for more info. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec choice
This is at least the third post under the subject 'Codec Choice' by the same sender. Why don't you stay within your first thread? Does posting over and over again increases chances of getting a solution? If so, then maybe I should try the same, as seems like an increasing trend on this list. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 7:13 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, Group () and Group_Count () will need to be used on certain extension. What if there are lot of clients on the kit with different routings some going to dahdi and some to different sip interconnects, how can we do it on whole kit basis. Or let me know if there is any other way to use these functions to achieve this. Thanks, D -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring and how about their stability and reliability? Or should I stay with 1.4? Sincerely, Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Include and Realtime
I think you asked this question earlier and there were good responses to it. There is nothing more to it than what people already suggested. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 8:56 AM, Dan Journo d...@keshercommunications.com wrote: Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include = client1_internal include = client1_outgoing_calls include = test_calls include = parkedcalls [client2_phones] include = client2_internal include = client2_outgoing_calls include = test_calls include = parkedcalls I'm creating an application to allow a secretary to create new client accounts. It uses mysql and realtime, and I want to avoid changing the extensions.conf file. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Did you use VMWare's hypervisor? I have no experience with it but I'll be using Proxmox with no KVM, just OpenVZ because the server's processors don't support hardware virtualization. I have worked for someone before with Asterisk 1.4s running on Proxmox, and there was no issue regarding virtulization of asterisk. Plus I am not using DAHDI or PRI, just plain SIP and IAX. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 10:07 AM, Bruce Komito bru...@wpti.net wrote: We moved a 1.4 installation to a VMWare environment some time ago and it was fairly uneventful. Still, if it were me, I wouldn’t change too many things at once and I would first wait until what I currently run is stable under VM. Once stable, I wouldn’t hesitate to upgrade and that’s one of the nice things about running in a virtual environment. It’s makes upgrades such as that really easy, both from the standpoint of moving forward and reverting back, if necessary. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Tuesday, August 24, 2010 6:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Thanks for sharing your experience Bruce. I am going to use OpenVZ and hope it'll work fine. Is ESXi free or costs license, just in case OpenVZ won't work. The client I worked for, who was using OpenVZ had pretty moderately busy asterisk servers and didn't have any issues with it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 10:45 AM, Bruce Komito bru...@wpti.net wrote: We now run VMWare ESXi 4.0 on HP Proliant DL360 G5 and have not had any issues. A couple of years ago, we tried OpenVZ, but did not have good results. Don’t ask to me explain what the problem was, because that was the problem…we couldn’t figure it out. It was just unexplained erratic Asterisk behavior that we did not experience on dedicated hardware. And, we were not using any PRI or other boards…just plain old SIP and IAX. It could have been OpenVZ or it could have been something we did, but the result was the same. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Tuesday, August 24, 2010 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Did you use VMWare's hypervisor? I have no experience with it but I'll be using Proxmox with no... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?
Gorden, I agree with you and I moved to 1.4 only because I wanted to use the 'originate' command on asterisk CLI, and there was one more small little feature difference which I don't remember now, but nothing more than that, otherwise my 1.2 installation was just great. I know someone who didn't move from 1.0.9 for a long time as it was working just fine for his setup, and he had some serious call volume. Once I tried 1.6 and got in such a mess regarding the real-time that decided to roll back. I think I would prefer to keep 1.4 for production and install 1.6 or 1.8 for playing around with. That's the good thing about virtualization that I can install multiple servers not worrying about additional hardware or interference with rest of the setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-24 11:03 AM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Tue, 24 Aug 2010, Zeeshan Zakaria wrote: Hi list, I am planning a migration to virtual mac... Some of us are still using 1.2 because it's as stable and solid as it needs to be... Gordon -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
