Re: [asterisk-users] Asterisk Outlook integration
Hi BB, you could try this: http://asterisk-outlook-dialer.voip-singapore.qarchive.org/ Never tested it deeply but apparently seems to work fine. Giorgio Incantalupo Bruce B wrote: Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT even if contact exists. 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] Call forwrading but call transfer back
Do you forward the call from SIP phone or Asterisk dialplan. If it is from SIP Phone, above solution will not work. Infact any solution will not work except your softphone supports call forwarding based on some filter parameters. --AsteriskMan On 1/5/11, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Tuesday, January 04, 2011 9:36 AM To: asterisk-users Subject: [asterisk-users] Call forwrading but call transfer back Hi All, I have weird requirement for call forwarding. I have forward all call from A to B extension because A is very busy and sometime not available so B is taking care of all forwarding call from A. but in some case B need to transfer call to A and in this case call coming back to B again because of forwarding enabled. How to get rid on this condition ? How could B can transfer call to A ? Thanks, Satish This is a job for ex-girlfriend logic. Set up your dialplan like this (A=1001, B=1002) Exten = 1001,verbose(extension A-1001 handling) Exten = 1001,n,dial(SIP/1002) Exten = 1001/1002,n,dial(SIP/1001) If you dial 1001 from anywhere except 1002, you get sent to 1002. If you dial 1001 from 1002, you get sent to 1001. -- _ -- 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] 1.8 MIBs
On 110103 2154, Barry Miller wrote: On Mon, Jan 03, 2011 at 08:04:48PM -0800, Kirill Katsnelson wrote: Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? They're now part of the wiki. Thanks! -kkm -- _ -- 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] Do not disturbe
Flavio Miranda wrote:
I really would like to understand why dont works!
The variable should be ${CALLERID(num)}, unless you have a very old
version of Asterisk.
Doug
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Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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Re: [asterisk-users] Log and forward calls to cellphone?
Le 04/01/2011 20:50, Sebastian a écrit : Hi, On 01/04/2011 03:24 PM, Administrator TOOTAI wrote: Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery Any chance you could provide few more details please? Mainly which phone, what version of Android, and how many hours on standby when using OpenVPN. Also, which application were you running through OpenVPN and was it in constant use (the app). Hmmh, most of all those infos were given in the original message, see below ;-). HTC Hero rooted with Android 2.1 VillainRom9.0.0 Sip client is SipDroid (tested few others but never got them connecting to our Asterix). OpenVPN drains battery in less then 4 hours without calling. SipDroid is able to connect using 3G, I use it from time to time. How I use my mobile phone: . in the office, connected through WIFI with Asterisk server: can pass and receive calls, any technologie . out of the office: incoming calls to office numbers are routed to my mobile number after x seconds of no answer from the office phones. My mobile subscription include free calls to few landlines numbers 24h/24h 7d/7d: one of them is the office number. Calling this number give me an IVR from where I can enter the number I wish to call using our SIP routes. As I told, the best SIP client I had is Nokias one. Fully integrated, working out of the box. I am investigating using OpenVPN with Android - and I would find the above detail very useful. Many thanks, Sebastian [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- Daniel -- _ -- 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] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi_transcode.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko /lib/modules/2.6.26-2-686/dahdi/dahdi_echocan_kb1.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko
[asterisk-users] Calls Transfers
Hello! I am trying to figure out how call transfers work in SIP. What extension does the transferring and transferee devices go to? Elliot -- _ -- 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] Add Privacy: id to SIP-invite
Hello list, is it possible to add the field Privacy: id to a SIP INVITE message ? INVITE sip:32444666...@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP1 .2.3.4:5060 From: R321113 sip:3211133...@1.2.3.4;tag=2096790244 To: sip:32444666...@1.2.3.4 Call-ID: 3b040826e909d311880a009033060...@192.168.12.40 mailto:3b040826e909d311880a009033060...@192.168.12.40 CSeq: 34677 INVITE Contact: sip:32444666...@1.2.3.4:5060 Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE Content-Length: 203 Content-Type: application/sdp Max-Forwards: 69 Supported: replaces,answermode,100rel User-agent: (innovaphone IP800/6.00 sr2-hotfix16 [09-60901.35/424/110]) *Privacy: id* How can I do this in the Asterisk dialplan ?? SIPAddHeader ?? Kind regards, 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
Re: [asterisk-users] VoIP PoE phones for restaurant
It is indeed possible (quite common, actually) to run the wiring as you describe. If you want to keep the data and voice traffic separate, you can use VLANs to do so. Your switches will need to support VLANS, and you will need to configure VLANs to separate the voice and data traffic. As I understand it, though, you are still subject to the bandwidth limitations of the underlying network, so it's still possible that heavy traffic from the PC might affect the voice traffic. QOS or other methods might be used to help avoid this. For this reason, I personally prefer to keep my voice and data LANs physically separated when possible. Obviously, cost and complexity do increase somewhat. It's probably not a good solution for everyone, but it sounds like you have a pretty small installation and you might decide that the additional cost is justified. Tom -- Tom, amazing suggestion. I have been on the fence on how I should do this, and your last paragraph succinctly outlines what I've been thinking and leaning towards. I will follow your direction. Thank you for your response. I'm good at being molded. -Andy -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. What is this dream phone you speak of? Please help me in located it. I don't want to make a mistake with purchasing the wrong thing. I've never seen such a thing. We've got two noisy kitchens that need to talk back and forth. This is what I first imagined I would find, but I've not found this yet. Thank you for your response Tilghman. -Andy -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
On 01/04/2011 09:02 PM, mgra...@mstvp.com wrote: IMHO G.722 beats Clarity By Polycom every time. I had an IP335 for review before they launched. The audio quality is the same as the better models (IP450/550/650) only the user interface is different. Very good speakerphone, too. Review here: http://www.mgraves.org/2010/01/review-polycom-soundpoint-ip335-entry-level-hdvoice-ip-phone/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Michael, thanks for your response and sharing your excellent review! Beautiful website btw; I like the color scheme. -Andy -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
I would. The whole Polycom line seems designed for desktop use, and the speakers just don't get very loud. I have especially had this complaint about the ring volume, even at some desktops! In the hotels where we have installations that include busy kitchen extensions there seems to be no substitute for an old analog wall mount phone with a really loud ringer (backed by an ATA). That doesn't help you with intercom though... j Jeff, thank you for your insight. Thats the second vote that I shouldn't be getting a regular phone to act as an intercom in a kitchen. -Andy -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering 14kHz bandwidth. These are most often found in Polycom phones, but they are available elsewhere. The only widely supported HD codec is G.722. Pretty much anything offering wideband voice supports G.722. Steve -- _ -- 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] [tech] VoIP PoE phones for restaurant (kitchen)
On 01/05/2011 07:50 AM, Andy Graybeal wrote: I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. What is this dream phone you speak of? Please help me in located it. I don't want to make a mistake with purchasing the wrong thing. I've never seen such a thing. We've got two noisy kitchens that need to talk back and forth. This is what I first imagined I would find, but I've not found this yet. Thank you for your response Tilghman. -Andy I've found this: http://www.888voipstore.com/cyberdata-voip-intercom-with-keypad-011078.html It's really expensive! $450 for one intercom! What do you guys think of this, and has anyone used such a thing? -Andy -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/05/2011 01:29 AM, Bruce B wrote: Hi Everyone, You should really spend some time learning how to use a widely available search engine, such as Google or Bing. Many of the questions you ask here could be quickly answered that way. 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? That depends on what you mean by the G.722 HD voice codecs. G.722 (no suffix or annex) is a specific voice codec, and it is unrelated to Siren7 and Siren14. G.722.1 is the same thing as Siren7, and G.722.1 Annex C is the same thing as Siren14. These are fairly well explained on the Wikipedia pages for G.722 and G.722.1. 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? There is no the HD voice codec. The most widely available HD voice codec is G.722, but there are others available as well in endpoints from various manufacturers. G.722.1 and G.722.1C are available in many Polycom SIP devices, but not yet widely available in other devices (although there are softphones that have them), although they are available as binary add-on modules for Asterisk. 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Which two are you referring to here? G.722 is frequently used over the Internet (the weekly VUC makes it available, for example), and G.722.1/C use less network bandwidth than it does, so they should be usable in the same situations as well. Whether they are advisable for you to use or not depends entirely on your network connectivity, bandwidth and packet loss situation. None of them use more network bandwidth than G.711, though, so if your network connections can already handle G.711 calls without causing unacceptable audio disturbance, you should be fine using any of these codecs as well. -- 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
Re: [asterisk-users] Calls Transfers
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Wednesday, January 05, 2011 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calls Transfers Hello! I am trying to figure out how call transfers work in SIP. What extension does the transferring and transferee devices go to? Elliot A call transfer is not a SIP/DAHDI or any other type of technology/branch function. A call transfer is simply the reassignment of leg B of a call to a new leg B. If I call from SIP/100 to SIP/101 and SIP/101 transfers me to DAHDI/5551212, the same actions take place as if they had sent me to SIP/102. -- _ -- 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] Post Dial Delay + Playtones
Can anyone give me some pointers on the following, in our setup (ast 1.6.3) we use international carriers to terminate calls for a callingcard system, we have an issue where there can be a very long delay after dialing but before the far end begins to ring. I would like to play a tone every second during this period (before ringing) but then cut off the tone once the far end either sends ringing or progress in band (183). Iv tried using Progress() and Playtones() before the dial but Playtones cuts off as soon as i hit Dial. Iv tried using the Sip/xxxSip/xxx trick to get Playtones to continue (which kinda works) unfortunately this also over writes the far end ringing, so the user only hears the tone every second until the far end answers. I have tried a lot of things but can't seem to get the exact behaviour we want. Ask about our great deals on mobile phone contracts from the leading networks. We also provide high speed unlimited usage Business Broadband from £15.99 per month. Contact your account manager for further details. VoIPtalk network on Twitter. Get the latest news, service information, rates and much more - http://www.twitter.com/voiptalktweets. Get latest status messages on http://www.twitter.com/voiptalkstatus This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. -- _ -- 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] problems inserting dahdi modules using Debian Leni
On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote: Shaun Ruffell sruff...@digium.com wrote: On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko snip and []# dmesg -c /dev/null; modprobe dahdi; dmesg; lsmod | grep dahdi FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Unknown symbol in module, or unknown parameter (see dmesg) [25991.968325] dahdi: no symbol version for crc_ccitt_table
Re: [asterisk-users] Add Privacy: id to SIP-invite
Jonas
This is how we are doing it.
exten = s,n,SipAddHeader(P-Asserted-Identity: :${siteDefaultCIDNumber})
exten = s,n,GosubIf($[${gbl_CallPrivacy}=id]?rfc-3325-CPN,1)
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:
,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}...@${from_ip}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
;exten = rfc-3325-CPN,n,SetCallerPres(prohib_not_screened) ; this might
not be needed --- needs further testing
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
Good Luck
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 Good LuckBryant Zimmerman (ZK Tech Inc.)616-855-1030
Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, January 05, 2011 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Add Privacy: id to SIP-invite
Hello list,
is it possible to add the field Privacy: id to a SIP INVITE message ?
INVITE sip:32444666...@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP1 .2.3.4:5060
From: R321113 sip:3211133...@1.2.3.4;tag=2096790244
To: sip:32444666...@1.2.3.4
Call-ID: 3b040826e909d311880a009033060...@192.168.12.40
CSeq: 34677 INVITE
Contact: sip:32444666...@1.2.3.4:5060
Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPD
ATE
Content-Length: 203
Content-Type: application/sdp
Max-Forwards: 69
Supported: replaces,answermode,100rel
User-agent: (innovaphone IP800/6.00 sr2-hotfix16 [09-60901.35/424/110])
Privacy: id
How can I do this in the Asterisk dialplan ?? SIPAddHeader ??
Kind regards,
Jonas.
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Re: [asterisk-users] Log and forward calls to cellphone?
Hi, On 01/05/2011 10:49 AM, Administrator TOOTAI wrote: Le 04/01/2011 20:50, Sebastian a écrit : Hi, On 01/04/2011 03:24 PM, Administrator TOOTAI wrote: Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery Any chance you could provide few more details please? Mainly which phone, what version of Android, and how many hours on standby when using OpenVPN. Also, which application were you running through OpenVPN and was it in constant use (the app). Hmmh, most of all those infos were given in the original message, see below ;-). HTC Hero rooted with Android 2.1 VillainRom9.0.0 Sip client is SipDroid (tested few others but never got them connecting to our Asterix). OpenVPN drains battery in less then 4 hours without calling. SipDroid is able to connect using 3G, I use it from time to time. How I use my mobile phone: . in the office, connected through WIFI with Asterisk server: can pass and receive calls, any technologie . out of the office: incoming calls to office numbers are routed to my mobile number after x seconds of no answer from the office phones. My mobile subscription include free calls to few landlines numbers 24h/24h 7d/7d: one of them is the office number. Calling this number give me an IVR from where I can enter the number I wish to call using our SIP routes. As I told, the best SIP client I had is Nokias one. Fully integrated, working out of the box. Thanks very much for the above info. Sebastian I am investigating using OpenVPN with Android - and I would find the above detail very useful. Many thanks, Sebastian [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- _ -- 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 SIP, multiple AX servers question
On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote: It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. We have one table which is serving both purposes (peers and reg). When we want to route a call to an ATA, we first look up that ATA's regserver in that table, and then construct a SIP URI based upon that regserver address. In that way, we route the call through the server to which the ATA is currently registered. So I guess we're covered already in the scenario you describe. It seems like not a great design to have to have a private sipregs table for every server in our pool, especially given that the pool will grow (or maybe shrink) over time. Is that really the recommended design? I haven't seen any articles describing that setup for RealTime in a multi-server environment. Thank you, Bryan -- _ -- 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] Blind Transfer not working - 1.4.38
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17
So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.
I ran a SIP trace of this happening and it appears to be attempting to
do the transfer:
-
--- (12 headers 0 lines) ---
Call 7c5d5a603b2803fd7e451de82...@x.x.x.x got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 5...@pack-local by pack...@domain.co.uk
--- Transmitting (NAT) to x.x.x.x:52753 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
From: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
To: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
Call-ID: 7c5d5a603b2803fd7e451de82...@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:incoming mobile number@x.x.x.x
Content-Length: 0
set_destination: Parsing sip:pack...@192.168.1.105:3072;line=guuuyf05 for
address/port to send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:pack...@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
To: sip:pack...@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
Contact: sip:incoming mobile number@x.x.x.x
Call-ID: 7c5d5a603b2803fd7e451de82...@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: incoming mobile number sip:incoming mobile
number@x.x.x.x;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
___
But as stated above, extension 504 doesn't ring and the call dies.
Now 504 is a valid extensions in the context pack-local
select * from extensions where exten='_5XX';
+---++---+--+---+---+
| id| context| exten | priority | app | appdata
|
+---++---+--+---+---+
| 65127 | pack-local | _5XX |1 | Macro |
stdexten|${EXTEN}|pack-local|PACK |
+---++---+--+---+---+
Also, attended transfers work without a problem.
Both SIP phones used were Snom phones.
Has anyone encountered an issue like this before?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
If you do get a Polycom, the old 501 (discontinued) have a louder ring (or can be configured to have a louder ring, don`t quite remember) then the newer ones. But the others are right: it's not meant for this, at least not in a noisy environment. What can work though is a Polycom 321, with a (loud) speaker plugged into the 3.5mm port and properly configured to have the speaker take the call (see paging app and Polycom admin manual). It`s a bit of a hassle but it`s much better than the unreliable and expensive Cyberdata paging products (I hated the one I tried, replaced it with a 321 as described). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal Sent: Wednesday, January 05, 2011 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; t...@casanueva.com Subject: Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen) I would. The whole Polycom line seems designed for desktop use, and the speakers just don't get very loud. I have especially had this complaint about the ring volume, even at some desktops! In the hotels where we have installations that include busy kitchen extensions there seems to be no substitute for an old analog wall mount phone with a really loud ringer (backed by an ATA). That doesn't help you with intercom though... j Jeff, thank you for your insight. Thats the second vote that I shouldn't be getting a regular phone to act as an intercom in a kitchen. -Andy -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/05/2011 07:07 AM, Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. Asterisk actually supports both the Siren* and G.722.1* names in SDP negotiations. I wasn't aware there were bitstream incompatibilities between the Siren* and G.722.1* variants, even though the code may be slightly different... so Asterisk uses a single codec module for both variants. -- 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
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. The Siren7 and Siren14 codecs in Asterisk are licensed code from Polycom, so they are indeed the Siren7 and Siren14 codecs. They will interoperate with any other vendor who has licensed those codecs from Polycom. -- Tilghman -- _ -- 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] DTMF-troubles with Snom
Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for 2 : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan 5 17:06:39] VERBOSE[29172] pbx.c: [Jan 5 17:06:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test1-0701, 15) in new stack *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* What follows is a prompt again, and it automatically chooses option 2 : [Jan 5 17:06:41] VERBOSE[29172] file.c: [Jan 5 17:06:41] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') *[Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701* Even without pressing 2 on the Snom phone, option 2 is chosen in the menu. The above is different when I do the same with a Grandstream device : [Jan 5 17:14:15] VERBOSE[29384] file.c: [Jan 5 17:14:15] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' ( language 'nl') [Jan 5 17:14:17] VERBOSE[29384] pbx.c: [Jan 5 17:14:17] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' *[Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714* [Jan 5 17:14:38] VERBOSE[29384] file.c: [Jan 5 17:14:38] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:14:39] VERBOSE[29384] pbx.c: [Jan 5 17:14:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack *[Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714*** Here I explicitly chose option 2 by pressing on button 2. What is going on with the Snom ? There is a difference in duration (160ms vs 100ms). Is that the problem ?? Kind regards, 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
[asterisk-users] Polarity Reverseal....with analog line
Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'-- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...-- Executing [...@from-pstn:1] Answer(Zap/5-1, ) in new stack-- Executing [...@from-pstn:2] Playback(Zap/5-1, vm-intro) in new stack-- Zap/5-1 Playing 'vm-intro' (language 'en')[Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 5-- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'-- Hungup 'Zap/5-1' I am using 1.4.30 and zaptel 1.12. Any cluess?*---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
On Wednesday 05 January 2011 06:50:19 Andy Graybeal wrote: I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. What is this dream phone you speak of? Please help me in located it. I don't want to make a mistake with purchasing the wrong thing. I've never seen such a thing. We've got two noisy kitchens that need to talk back and forth. This is what I first imagined I would find, but I've not found this yet. Top link on Google for stainless steel SIP intercom: http://www.adamtelco.com/valcom-vip-172l-st-stainless-steel-sip-intercom- doorphone.html Cyberdata appears to have another, too: http://www.alloy.com.au/010935.htm Yet another: http://www.zenitel.com/en/Stentofon/Products/Tamper--Vandal-Resistant- Substations/SIP-Vandal-Resistant-Substation/ -- Tilghman -- _ -- 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] TE420 issue: card 0 span N: isr2=XX isr3=Y
I have just built a new system using an HP DL360G7 with a TE420 T1 card, and this is the first system using a generation 7 server. I'm not sure whether that is an issue or not. I am using Asterisk 1.2, and Zaptel 1.4.12.1 with patches for GEN5 of the TE420 card. I have successfully used this combination on several systems based on the DL360G6 and TE420(gen5), which have been in production for many months now. However, on the new server, while doing loopback testing, I have found that several minutes after starting Zaptel and Asterisk, I start to get several console and syslog messages per second of the form: card 0 span N: isr2=XX isr3=Y where N can be 0, 1, 2 or 3; XX can be 03, 40, 80, 83, C0 or C3; Y is usually 0 or 2, but occasionally 3. I have no idea what these numbers mean, except that in the code, the message is output if isr2 is non-zero or if isr38 is non-zero. I don't know whether this behaviour is caused by cross-connecting the ports using T1 crossovers (port 1 to 3 and 2 to 4, with 3 and 4 set to pri_net instead of pri_cpe), and would disappear when connecting to real T1 lines, or is caused by the new hardware. Once the messages have started, they continue even if I stop Asterisk; in order to stop them I need to shut down Zaptel too. Originally I had the following in zaptel.conf: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs And then, wondering if it was to do with timing slips, tried this: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs That didn't help, so I then tried this: span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs That seems to have helped at the moment, but I don't know whether that is coincidence or to be expected. Any advice would be greatly appreciated! I'm not able at present to move to DAHDI or a newer Asterisk. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
On 01/05/2011 7:50 AM, Andy Graybeal wrote: We've got two noisy kitchens that need to talk back and forth. Andy, Why, exactly, are you trying to combine an inter-kitchen intercom and your phone system? Might it make more sense to have a non-phone-based intercom system, plus a phone for making phone calls? Tom -- _ -- 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] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote: Shaun Ruffell sruff...@digium.com wrote: On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko snip and []# dmesg -c /dev/null; modprobe dahdi; dmesg; lsmod | grep dahdi FATAL: Error
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote: Shaun Ruffell sruff...@digium.com wrote: On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko snip and []# dmesg -c /dev/null; modprobe dahdi; dmesg; lsmod | grep dahdi FATAL: Error
Re: [asterisk-users] Calls Transfers
Thanks! Although there is no difference between SIP or any other technology, how does Asterisk reconcile the channel variables? For example: 1. A calls B 2. B answers 3. B transfers A to C 4. C picks up call Now, when B makes a transfer (say by pressing the transfer button on a sip phone), what channel variables and contexts are being used? Does Asterisk take all of Channel variables (context, accountcode, etc.) that would normally be assigned to B and apply that to the channel A is currently using or is something more sophisticated going on? Thanks, Elliot On Wed, Jan 5, 2011 at 4:08 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Wednesday, January 05, 2011 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calls Transfers Hello! I am trying to figure out how call transfers work in SIP. What extension does the transferring and transferee devices go to? Elliot A call transfer is not a SIP/DAHDI or any other type of technology/branch function. A call transfer is simply the reassignment of leg B of a call to a new leg B. If I call from SIP/100 to SIP/101 and SIP/101 transfers me to DAHDI/5551212, the same actions take place as if they had sent me to SIP/102. -- _ -- 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] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote: Shaun Ruffell sruff...@digium.com wrote: On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencermarks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) And with the crc_ccitt module loaded you still cannot run modprobe dahdi? If so, what is the output of: []# cat /lib/modules/`uname -r`/modules.dep | grep dahdi.ko: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko snip and []# dmesg -c /dev/null; modprobe dahdi; dmesg; lsmod | grep dahdi FATAL: Error
Re: [asterisk-users] Realtime SIP, multiple AX servers question
On 01/05/2011 09:39 AM, Bryan Field-Elliot wrote: We have one table which is serving both purposes (peers and reg). When we want to route a call to an ATA, we first look up that ATA's regserver in that table, and then construct a SIP URI based upon that regserver address. In that way, we route the call through the server to which the ATA is currently registered. So I guess we're covered already in the scenario you describe. It seems like not a great design to have to have a private sipregs table for every server in our pool, especially given that the pool will grow (or maybe shrink) over time. Is that really the recommended design? I haven't seen any articles describing that setup for RealTime in a multi-server environment. Asterisk Realtime was designed to make dynamic configuration possible, and then later extended to provide a way to store *some* data outside of astdb. It was never intended to provide a failover or data-sharing mechanism, and none of the code attempts to take that into account. -- 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
Re: [asterisk-users] Realtime SIP, multiple AX servers question
On Wednesday 05 January 2011 09:39:00 Bryan Field-Elliot wrote: On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote: It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. We have one table which is serving both purposes (peers and reg). When we want to route a call to an ATA, we first look up that ATA's regserver in that table, and then construct a SIP URI based upon that regserver address. In that way, we route the call through the server to which the ATA is currently registered. So I guess we're covered already in the scenario you describe. It seems like not a great design to have to have a private sipregs table for every server in our pool, especially given that the pool will grow (or maybe shrink) over time. Is that really the recommended design? I haven't seen any articles describing that setup for RealTime in a multi-server environment. Sorry, but a private table for sipregs for each server was exactly what it designed for, in order to separate out values which change per-server from general configuration (same for every server). While I understand that you're presently using a separate lookup into that table, DUNDi is the (scalable!) protocol meant to perform this task for you. Clearly, using a shared sipregs table has its own set of problems; rather than sticking to your flawed configuration, I would think that you would jump at the chance to fix it. -- Tilghman -- _ -- 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] Asterisk replying to wrong port for NOTIFY messages
See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James --- SIP read from zzz.zzz.zzz.44:9363 --- NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: xxx-xxx- sip:xxx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M Max-Forwards: 70^M Contact: xxx-xxx- sip:xx...@192.168.1.140:9363^M Event: keep-alive^M User-Agent: Cisco/SPA509G-7.4.6-0002fdff90a4^M Content-Length: 0^M ^M - [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- (11 headers 0 lines) --- [Jan 5 13:46:36] VERBOSE[3919] logger.c: --- Transmitting (no NAT) to zzz.zzz.zzz.44:1025 --- SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3;received=zzz.zzz.zzz.44^M From: xxx-xxx- sip:xx...@pbx1.mydomain.com;tag=467525dd6fac949do0^M To: sip:pbx1.mydomain.com;tag=as0493c604^M Call-ID: 707176dd-38f47...@192.168.1.140^m CSeq: 118907 NOTIFY^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M Supported: replaces^M Content-Length: 0^M -- _ -- 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] Polarity Reverseal....with analog line
Looks like your telco is sending you polarity reversal on sending you a call. Which is one of the types of setups for analog lines.l From your console output it looks like the call was handled just fine other than the 'weird event' notification, which I'm not familiar with. On 01/05/2011 11:50 AM, Edwin Quijada wrote: Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1' [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-pstn:1] Answer(Zap/5-1, ) in new stack -- Executing [...@from-pstn:2] Playback(Zap/5-1, vm-intro) in new stack -- Zap/5-1 Playing 'vm-intro' (language 'en') [Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 5 -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I am using 1.4.30 and zaptel 1.12. Any cluess? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
Top link on Google for stainless steel SIP intercom: http://www.adamtelco.com/valcom-vip-172l-st-stainless-steel-sip-intercom- doorphone.html Cyberdata appears to have another, too: http://www.alloy.com.au/010935.htm Yet another: http://www.zenitel.com/en/Stentofon/Products/Tamper--Vandal-Resistant- Substations/SIP-Vandal-Resistant-Substation/ Tilghman, Thank you for the response. The zenitel.com link looks nice in the picture! -Andy -- _ -- 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] Weird phone behavior after recent CentOS 5 update
For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to voice mail. But there is no audible 'ring' to the caller. Just nothing - blank, empty silence. Of course any automated answering system (ie. business phone menu, etc.) that we have works just fine. Its just the lines that go directly to an internal phone that are no longer providing any audible ring which is sending a message to the caller that their call didn't go through. Does anyone have any idea what might cause this? Myles -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 01:04 AM, Tilghman Lesher wrote: On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. The Siren7 and Siren14 codecs in Asterisk are licensed code from Polycom, so they are indeed the Siren7 and Siren14 codecs. They will interoperate with any other vendor who has licensed those codecs from Polycom. What Polycom licence to everyone is actually G.722.1 and G.722.1C. Been there. Done that. Steve -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 12:05 AM, Kevin P. Fleming wrote: On 01/05/2011 07:07 AM, Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. Asterisk actually supports both the Siren* and G.722.1* names in SDP negotiations. I wasn't aware there were bitstream incompatibilities between the Siren* and G.722.1* variants, even though the code may be slightly different... so Asterisk uses a single codec module for both variants. I am unclear how compatible or incompatible the bitstreams may be. What I know (from implementing these codecs) is that the source code Polycom provide licencees, as the basis for developing their own G.722.1 and G.722.1C codecs, has several comments referring to things not being quite the same as Siren7/Siren14. However, they don't hand out the actual Siren7/Siren14 source code, so I don't know how much divergence there is. Steve -- _ -- 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] Weird phone behavior after recent CentOS 5 update
Some more info on this weirdness Seems that if I play any audio out from Asterisk first the problem goes away. its almost like the entire audio engine isn't being 'initialized' or something on those direct calls. I found that if I execute a Swift(Please Wait) before I call the Dial command in my extensions, then it plays the ringing tone as expected. But if I receive a call, and then try and have it transferred directly to a phone extension via a Dial command (which used to work perfectly well), the calling party never hears a ringing tone. But if I play some audio purposely before, then it rings Myles -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
On 01/05/2011 01:51 PM, Tom Rymes wrote: On 01/05/2011 7:50 AM, Andy Graybeal wrote: We've got two noisy kitchens that need to talk back and forth. Andy, Why, exactly, are you trying to combine an inter-kitchen intercom and your phone system? Might it make more sense to have a non-phone-based intercom system, plus a phone for making phone calls? Tom -- _ -- 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 Phones make great intercoms when the volume gets loud enough. The polycom 321/331 doesn't have a very loud speakerphone. You may be interested in a paging system. -- _ -- 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
