[asterisk-users] audio disappeared after ztdummy install
Hi All, Can't explain what happened, last night i was setting the voicemail configuration, and it worked properly: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0, @VM-1000) in new stack -- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') i can hear the audio playing here. earlier i started playing with meetme, and since i don't have any zap cards, i chose to use ztdummy, -- Executing [EMAIL PROTECTED]:1] MeetMe(SIP/1000100-08206da8, 6000) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- SIP/1000100-08206da8 Playing 'conf-getpin' (language 'en') -- SIP/1000100-08206da8 Playing 'conf-onlyperson' (language 'en') from that message asterisk is playing conf-getpin, so i entered my conference pin number, even though i don't hear any audio, then it tried to play conf-onlyperson, still i dont hear anhything. then i tried my voicemail retrieval -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000101-0822b6c0, @VM-1000) in new stack -- SIP/1000101-0822b6c0 Playing 'vm-login' (language 'en') same thing it's playing something but i don't hear anything. i tried playing around with my codecs, i even downloaded the alaw core and extra sound files. what do you guys think happened? it was working before i enabled ztdummy. i tested disabling the ztdummy then i can hear the audio at the voicemail but conference of course does not work now. i'm using zaptel-1.4.9.2, i tried downgrading to 1.4.8 down to 1.4.7. but still the same issue. Regards, Nhadie - Never miss a thing. Make Yahoo your homepage.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many maximum SIP Registrations can Asterisk Handle
Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, - Abid Saleem Choudhary Team Lead VoIP Networks Comcerto Bahrain W.L.L. Direct: (973) 13301504 Mobile: (973) 36080504 Tel: (973) 13301100, Fax: (973) 13301101 MSN: [EMAIL PROTECTED] WebSite: http//Comcerto.net P.O. Box: 311100 Manama, Kingdom of Bahrain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many maximum SIP Registrations can Asterisk Handle
On Sun, Mar 30, 2008 at 8:31 AM, Abid Saleem Choudhary [EMAIL PROTECTED] wrote: Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Depends a lot on a few factors the ones I can think of off the top of my head are: 1. The registration interval you use, the smaller the interval the greater the load but the better at keeping NAT connections open, 2. Whether you use the qualify setting to help with keeping NAT connections open, 3. Whether you are using realtime or a configuration file. 4. The load from calls, With a best case scenario the answers to the above questions would be: 1. 3600, 2. No, 3. Configuration file, 4. Minimal With that set up I'd guess that somewhere between 5,000 and 10,000 registrations would be something your hardware could handle. However it would be normal that once the number of SIP accounts gets to a certain point it becomes to painful to configure them via a config file which means realtime with it's database option comes into play. At that point you are better off taking the load from registrations away from Asterisk. Asterisk is not a particularly good SIP Registrar anyway so putting in something like OpenSER is a superior option. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] breaking DNID into country code, area code, and local code
Dear friends, I am wondering if there is any efficient way of extract the country code, area code, and local code into 3 different variables from one DNID that can look like 001630233-4333 or 0086213345333? International code can be 011, or 00. National code can be 0 or 1 Country code can have 2 or 3 digits Area code can have 2 or 3 digits Local num can be 7-10 digits Is there anyway to break this down efficiently in the dialplan or AGI? Any comment will be greatly appreciated. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] breaking DNID into country code, area code, and local code
Am Sonntag, den 30.03.2008, 16:56 +0800 schrieb mark morreny:
Dear friends,
I am wondering if there is any efficient way of extract the country
code, area code, and local code into 3 different variables from one
DNID that can look like 001630233-4333 or 0086213345333?
International code can be 011, or 00.
National code can be 0 or 1
Country code can have 2 or 3 digits
Area code can have 2 or 3 digits
Local num can be 7-10 digits
Is there anyway to break this down efficiently in the dialplan or AGI?
I think it can not be done efficiently, reliably, and for international
numbers.
The first problem would be to create the uniform international number in
+(X[XX])[YYY] format.
For example consider the number
01149228730
This might very well be a valid Sheffield, UK, number (no idea if it is,
and I will not call to find out :-) of area code 0114 and local
seven-digit number 9228730.
If dialled from US it will connect you to the University switchboard in
Bonn, Germany. (I had to find a really short number to fit the
seven-digit dialplan of Sheffield).
The problem is that some countries have 011 being (part of) a valid
area code, banning it as identification for this is an international
number dialled from North America. Vice versa some countries seem to
have valid uses for 00 that mean different things than international
dialling. I think it was used for operator in Spain back when they
had 07 for international dialling, and had been in some area codes in
Russia until they decided to migrate from 8~10 to 00 for international
dialling until 2010.
So getting your numbers standardized to + C[CC] A[] SSS[SS]
may already break on those problems.
Sorry, but you are not all happy either once you have that standardized
form. US is easy with the fixed +C AAA SSS form, and some countries
are similarly easy as they have fixed-length area codes (France, AFAIK)
or no area codes at all (Denmark). UK has two (London 20, Coventry 24
and a few others) up to four (afaik) area code digits, which possibly
can be recignized by logic, as +44 2 always is two-digit, and +44 1x1
and +44 11x are always three-digit - I do not know if that is valid
universally though. Any logic breaks when it comes to German area
codes, where +49 x0 may or may not be a valid area (30 - Berlin, but
5031 - Wunstorf, and 209 - Gelsenkirchen), and area codes range from
two to five digits, with a few three-digit subsribers nearly anywhere,
but up to nine digit subscriber numbers in Berlin.
For some countries information may even be hard to get - although you
probably will not receive many calls from Benin, Ethiopia or Mongolia,
and if you do indeed, you will have no trouble getting their local
telephone system explained.
Once you have your numbers standardized ($NUMBER in +xxx form) you
could of course query a database, looking for ${NUMBER:1:7} down to
${NUMBER:1:2}, such that if applicable the Country-/-Area can be
returned as string, as a fall back the country only, and if nothing
helps, the number can be discarded as invalid (assuming you have a
complete list of country prefixes).
I think you will not find anything much simpler that which can handle
the structure of phone numbers, as that is for historical and political
reason rather messy ;-)
BR
Anselm
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Re: [asterisk-users] How many maximum SIP Registrations can Asterisk Handle
I'd say several thousands i n normal circumstances, but i would also like to recommend not to use asterisk for large scale registrations, its better to do those on SER for example, and route the calls through asterisk for termination, voicemail, conferencing etc. It all depends on how fast the reregistration time is set on those end devices and how much the registrations will collide in the same small interval. SER doesn't handle audio so even if the registration gets a little delayed because a flood arrives, the audio won't suffer. Zoa Abid Saleem Choudhary wrote: Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, - Abid Saleem Choudhary Team Lead VoIP Networks Comcerto Bahrain W.L.L. Direct: (973) 13301504 Mobile: (973) 36080504 Tel: (973) 13301100, Fax: (973) 13301101 MSN: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] WebSite: http//Comcerto.net P.O. Box: 311100 Manama, Kingdom of Bahrain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] e164.org
On 00:36, Sun 30 Mar 08, Armin Schindler wrote: On Sat, 29 Mar 2008, Grey Man wrote: Does anyone know if the e164.org ENUM service is still active? If anyone who has anything to do with the e164.org ENUM site monitors this list could you check your signup page as the Captcha's (the test to see if you are human) fails for both the text and audio tests every time. I'd post a message on the e164.org forums but the signup page there has the test missing altogether. I don't really know the 'official' status, but I use it and it does work without problems. Same here -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Langugae issue
Hi list I add new directory for Arabic voices support and I 'd translated all the English voices files into Arabic , with language = ar ,and it is working fine ,except some problems in saying the number order ,because the Arabic structure is quite different for numbers ,where in English language we can say twenty two while the order should be two and twenty ,so please if you can guide me how to change the setting to do that . regads Ayman _ Watch “Cause Effect,” a show about real people making a real difference. Learn more. http://im.live.com/Messenger/IM/MTV/?source=text_watchcause___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Langugae issue
Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. At 09:20 AM 3/30/2008, aymen warfalli wrote: Hi list I add new directory for Arabic voices support and I 'd translated all the English voices files into Arabic , with language = ar ,and it is working fine ,except some problems in saying the number order ,because the Arabic structure is quite different for numbers ,where in English language we can say twenty two while the order should be two and twenty ,so please if you can guide me how to change the setting to do that . regads Ayman -- Watch Cause Effect, a show about real people making a real difference. http://im.live.com/Messenger/IM/MTV/?source=text_watchcauseLearn more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
This is a reasonably common problem. ztdummy uses the Linux kernel Real Time Clock (RTC) and something is wrong with it. The solution is to recompile your kernel, you should search the mailing list archives. Prepend site:lists.digium.com to your Google search to limit your search to the mailinglist archives. ronald ramos wrote: Hi All, Can't explain what happened, last night i was setting the voicemail configuration, and it worked properly: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0, @VM-1000) in new stack -- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') i can hear the audio playing here. earlier i started playing with meetme, and since i don't have any zap cards, i chose to use ztdummy, -- Executing [EMAIL PROTECTED]:1] MeetMe(SIP/1000100-08206da8, 6000) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- SIP/1000100-08206da8 Playing 'conf-getpin' (language 'en') -- SIP/1000100-08206da8 Playing 'conf-onlyperson' (language 'en') from that message asterisk is playing conf-getpin, so i entered my conference pin number, even though i don't hear any audio, then it tried to play conf-onlyperson, still i dont hear anhything. then i tried my voicemail retrieval -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000101-0822b6c0, @VM-1000) in new stack -- SIP/1000101-0822b6c0 Playing 'vm-login' (language 'en') same thing it's playing something but i don't hear anything. i tried playing around with my codecs, i even downloaded the alaw core and extra sound files. what do you guys think happened? it was working before i enabled ztdummy. i tested disabling the ztdummy then i can hear the audio at the voicemail but conference of course does not work now. i'm using zaptel-1.4.9.2, i tried downgrading to 1.4.8 down to 1.4.7. but still the same issue. Regards, Nhadie Never miss a thing. Make Yahoo your homepage. http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
On Sun, Mar 30, 2008 at 12:22:43AM -0700, ronald ramos wrote: Hi All, Can't explain what happened, last night i was setting the voicemail configuration, and it worked properly: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0, @VM-1000) in new stack -- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') i can hear the audio playing here. earlier i started playing with meetme, and since i don't have any zap cards, i chose to use ztdummy, -- Executing [EMAIL PROTECTED]:1] MeetMe(SIP/1000100-08206da8, 6000) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- SIP/1000100-08206da8 Playing 'conf-getpin' (language 'en') -- SIP/1000100-08206da8 Playing 'conf-onlyperson' (language 'en') from that message asterisk is playing conf-getpin, so i entered my conference pin number, even though i don't hear any audio, then it tried to play conf-onlyperson, still i dont hear anhything. then i tried my voicemail retrieval -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000101-0822b6c0, @VM-1000) in new stack -- SIP/1000101-0822b6c0 Playing 'vm-login' (language 'en') same thing it's playing something but i don't hear anything. i tried playing around with my codecs, i even downloaded the alaw core and extra sound files. what do you guys think happened? it was working before i enabled ztdummy. i tested disabling the ztdummy then i can hear the audio at the voicemail but conference of course does not work now. i'm using zaptel-1.4.9.2, i tried downgrading to 1.4.8 down to 1.4.7. but still the same issue. What's the output of 'zttest -v -c 2' ? What do you see in /proc/zaptel/1 when you use a newer version of zaptel (= 1.4.8) ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Langugae issue
Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal: Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. Dear Mike, for me it seems that this is what say.conf is good for: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/say.conf.sample?revision=105596view=markup (which seems to be considered the new format). Perhaps it would be better to implement Arabic there than by means of an AGI script. Be sure to check with the developers wether this will be relevant for Asterisk 1.4 or if you need to go with 1.6 SVN to benefit. Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Langugae issue
Tzafrir, Anselm, and others. Thanks for your comments on my suggestion to Ayman. As one who is familiar, but not-native speaker with Arabic, Hebrew, and several other classical Semitic family languages, it would require much more time to try to fit those into the linear structure of SAY.CONF than to deal with it in a directly parsed manner. I can say the same for some Asian languages too. The results would recognized but would not be culturally acceptable. OFF TOPIC COMMENTS: I am constantly amazed at cross-language translations that try to follow the western language standards in computerized applications. Historically, the use of numbers came relatively late to western languages. While I am proud to be an American (as well as a computer-geek), I have crossed the multi-lingual multi-cultural barriers many years back! END OFF TOPIC COMMENTS. ..mike.. At 11:23 AM 3/30/2008, you wrote: On Sun, Mar 30, 2008 at 05:16:00PM +0200, Anselm Martin Hoffmeister wrote: Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal: Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. Dear Mike, for me it seems that this is what say.conf is good for: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/say.conf.sample?revision=105596view=markup (which seems to be considered the new format). Perhaps it would be better to implement Arabic there than by means of an AGI script. Be sure to check with the developers wether this will be relevant for Asterisk 1.4 or if you need to go with 1.6 SVN to benefit. say.conf works nicely for some languages. I was not able to make something useful enough with its syntax for Hebrew, and from the little I know of Arabic syntax, it will share the same problem. One basic problem is that there's no gender-form parameter anywhere in the interface. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy / RTC error
It's taken me about a day just to get ztdummy to compile into a module that the amazon ec2 xen kernel will accept (you have to downgrade the version of gcc among other things), but now I'm getting the following error and I'm stumped: rtc: IRQ 8 is not free. WARNING: Error inserting rtc (/lib/modules/2.6.16.19-xen/kernel/drivers/char/rtc.ko): Input/output error ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control FATAL: Error inserting ztdummy (/lib/modules/2.6.16.33-xenU/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) Any idea where I can go from here? -- Drew Miller Iowa Democratic Party Information Technology Director Office: (515) 974-1682 Cell: (515) 451-4509 AIM: ItsDrewMiller MSN: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Langugae issue
On Sun, Mar 30, 2008 at 01:12:43PM -0400, Mike Trest - Personal wrote: Tzafrir, Anselm, and others. Thanks for your comments on my suggestion to Ayman. As one who is familiar, but not-native speaker with Arabic, Hebrew, and several other classical Semitic family languages, it would require much more time to try to fit those into the linear structure of SAY.CONF than to deal with it in a directly parsed manner. I can say the same for some Asian languages too. The results would recognized but would not be culturally acceptable. I may be mis-informed, but I believe quite a few western-european languages actually have exactly the same problem - the need to count in both a male and a female form. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with voicemail odbc
On Friday 28 March 2008 01:54:12 mark morreny wrote: I am still not able to store voicemail into mysql and I am hoping someone can help me out. snip There is no error coming out of asterisk. Can anyone please tell me what could be the problem? Just a thought, but have you selected the ODBC option in the Voicemail options within menuselect? If you haven't built ODBC support, that would explain why it does not store and does not complain. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note that you need to enable the various and random kernel flags to make this work, generally dealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. Norman Franke ASD, Inc. www.myasd.com On Mar 30, 2008, at 1:00 PM, [EMAIL PROTECTED] wrote: This is a reasonably common problem. ztdummy uses the Linux kernel Real Time Clock (RTC) and something is wrong with it. The solution is to recompile your kernel, you should search the mailing list archives. Prepend site:lists.digium.com to your Google search to limit your search to the mailinglist archives. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
On Sun, Mar 30, 2008 at 02:35:03PM -0400, Norman W. Franke wrote: All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. After doing this, I recompiled ztdummy and it worked. Note that you need to enable the various and random kernel flags to make this work, generally dealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms
Gonzalo Servat wrote: On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Any suggestions?? I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2 http://1.4.9.2. A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 http://1.4.9.2 ? Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2 http://1.4.9.2. As per your suggestion on IRC, I've checked out, compiled and installed Zaptel from SVN (1.4 branch). I reloaded the zaptel modules but ... no go. Do I need to recompile Asterisk too? Shouldn't it have picked up the alarm as a red alarm on the channel? I've no idea to be honest. (Besides the problem. Is 1.4 SVN recommended for that at the moment?) Also no idea. I was told to use 1.4 if I'm using Asterisk 1.6 so I went with that. - Gonzalo Try: /svn/zaptel/!svn/ver/3905/team/kpfleming/battery_alarms It worked for me. You should have to rebuild asterisk. We do need a new zaptel release though. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Alan Lord wrote: Lenz wrote: Hello list, after spending the best part of an afternoon trying to build Asterisk on an old EPIA VIA C3, I thought that writing a tutorial would make life easier for future compilers: http://astrecipes.net/index.php?n=356 I had never compiled Asterisk for a different architecture, and I'm pretty disappointed at how complex it is - building Zaptel, Libpri and Asterisk requires discovering three different procedures, and even passing the required architecture to the autoconfig module was not enough for a clean build - libpthread and libresolv would not link, so you have to add them manually. Aybody got an idea of who should be notified of this immediate problem, apart for the time-wasteful general compilation procedure? Thanks l. Hi there, I didn't find it too much trouble in a Via C700N system. But I wouldn't use one of the mainstream distros for the OS. They chew up system resources just trying to accommodate any hardware. The solution is to roll-your-own. See this series of articles on my blog... http://www.theopensourcerer.com/tag/asterisk/ The C7 supports full i686 features. The C3 is an older chip that is fully i586 and partially i686 compatible. If you have a distribution that is compiled with i586 optimizations, you won't have problems. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tests in VMWare
I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules from automatically loading in modules.conf, e.g. autoload=no, Asterisk will start. If I keep the default, autoload=yes, Asterisk fails to start (seg fault). I can't find in any of the other config files where Asterisk may be trying to load a module and therefore crashing the system. I'm really just trying to experiment with different features and configurations of multiple Asterisk machines and would prefer to do that in virtual space. I'm willing to make my configs available. I just thought I'd drop this email on the list hoping for the chance that someone has dealt and corrected this problem. :-) Thanks much in advance. -- Ein Bielaczyc [EMAIL PROTECTED] NOTICE: This E-mail (including attachments) is covered by the Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is confidential and may be legally privileged. If you are not the intended recipient, you are hereby notified that any retention, dissemination, distribution or copying of this communication is strictly prohibited. Please reply to the sender that you have received the message in error, then delete it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
Hi, For now i just turned off acpi. and it works now. just dont know what's the connection of that though :-) i will try to do the things you guys suggested also when i get the chance, thanks for you help! regards, nhadie --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Mar 30, 2008 at 02:35:03PM -0400, Norman W. Franke wrote: All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. After doing this, I recompiled ztdummy and it worked. Note that you need to enable the various and random kernel flags to make this work, generally dealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Advice on queue setup needed please.
Hello, I have two sites. Both the sites will require queues for their own reason, own campaigns, etc. Like site1 would handle product1, product2, product3, while site2 can do product4, but can also do customer support for product1 and if anything, can transfer to site1's product1 queue (usually via SIP) Also, a few of these queues will require AgentLogin() type logging in, and some might not (and they can use regular either AQM or just be statically logged in cases where they might need to be logged into two queues) Currently I have sip peers setup in sip.conf where 1xx is for site1 and 2xx is for site2. I've made queues like product1, product2 and for example peer 121 has context=product1 in sip.conf. I also have [product1] in extensions.conf. If a call comes in via DID, I do a Goto(product1,100,1) where the call is sent from [incoming] to [product1] where it enters a queue(product1), and if 121 needs to transfer to say, exten = 7,1,Dial. which is in [product1], it transfers. That's the type of scenario that's real nice. But I have a few complications. I currently somehow have these queues setup, but I need to make myself clear on a few things. And maybe even need suggestions/advice. All the sip peers use eyebeam. When a new queue needs to be added, how can I simply re-assign the agents to new queues without having to make any changes on the eyebeams and only on the asterisk server? In this way, I just tell a team leader, Ok, agents 215 to 230 are configured to handle product4 queue. So they log in and when it prompts for username, punch in any of the ones between 215 and 230. This way all peers 215 and 230 do all product4 related stuff and not depend on what the device is configured. For example, if the device is configured as 245 and the agent logs in as 215, it should only take up all sip settings of 215 and do whats required rather than saying stuff like this extension is not in context test which 245 is set under. I'm not sure if anyone is following me because this completely clouded for me as well. But if someone is, can they please recommend a way for me to keep this mess organized? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio disappeared after ztdummy install
ronald ramos wrote: Hi, For now i just turned off acpi. and it works now. just dont know what's the connection of that though :-) i will try to do the things you guys suggested also when i get the chance, thanks for you help! regards, nhadie --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Mar 30, 2008 at 02:35:03PM -0400, Norman W. Franke wrote: All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. After doing this, I recompiled ztdummy and it worked. Note that you need to enable the various and random kernel flags to make this work, generally dealing with the high-performance timer. I enabled: HPET Timer Support Enhanced Real Time Clock Support HPET - High Precision Event Timer HPET Control RTC IRQ Allow mmap of HPET I'm not sure if you can eliminate some of those, but this works for me and is stable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This may help; http://www.mail-archive.com/[EMAIL PROTECTED]/msg10707.html -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tests in VMWare
If you leave all of the modules enabled, which one does it have a problem with? You should be able to run asterisk -vc to see where it stops loading. The last line or so should give you the module that it tried to load before it failed. Based on the last time I tried to install under Ubuntu, you're probably failing to load the Zap module. Since you're in a VM and it's unlikely that you're using Zap for anything, you can disable chan_zap.so and see if your Asterisk starts properly then. Cheers, AR On Sun, 2008-03-30 at 20:50 -0400, Ein Bielaczyc wrote: I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules from automatically loading in modules.conf, e.g. autoload=no, Asterisk will start. If I keep the default, autoload=yes, Asterisk fails to start (seg fault). I can't find in any of the other config files where Asterisk may be trying to load a module and therefore crashing the system. I'm really just trying to experiment with different features and configurations of multiple Asterisk machines and would prefer to do that in virtual space. I'm willing to make my configs available. I just thought I'd drop this email on the list hoping for the chance that someone has dealt and corrected this problem. :-) Thanks much in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Star Wars Echo Sound
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Schall wrote: They are all connected directly to the same switch which asterisk also connects into. Its a small office (6 people). So what is the difference between the end to end system for people who don't get it and people who do? If you get someone who does get it to use one of the phones from the people who don't get it, do they still get it (i.e. maybe the people who don't get it just aren't noticing it). What is the RTP packet size in both situations? Should be 20ms, but may be 30ms. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH8FOkDQNt8rg0Kp4RAnGgAJ9dZ6dHSP7diScGB2eh682qewzywgCgqMwh KhAdhCKn5D0qASM0y0MVFPA= =IUbW -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ping
Just a test -- Alexey mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
Could you elaborate a bit more on : For example, if I install zaptel from source, your support contract with them is void. Does this mean it is impossible to run Asterisk on Vendor Supported versions of RedHat or Suse ? Thanks Michiel van Baak wrote: On 02:34, Sat 29 Mar 08, Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. I choose Debian because I was already using it. And because there are people out there that can help me. I dont want the support from suse or redhat because they wont help me when running anything that's not in their repositories. For example, if I install zaptel from source, your support contract with them is void. I also really like the Open and Free mindset of Debian. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tests in VMWare
On Sun, Mar 30, 2008 at 08:50:10PM -0400, Ein Bielaczyc wrote: I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. What version of Ubuntu? What version of Asterisk? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
There are people who will support your Debian / Centos / whatever boxes. If it is OK to ask on a non-commercial list, do you have a list of reliable O/S support folks. By this I mean companies with a support staff, as opposed to a really bright and talented guy who does it between classes in school. Historically our projects were on big HP iron with HP-UX support from HP THX Tzafrir Cohen wrote: On Sat, Mar 29, 2008 at 02:34:36AM -0400, Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. What exactly is supported? Specifically, RHEL does not include Zaptel. And is not likely to include the kernel Zaptel modules until Zaptel comes closer to mainline kernel. SLES includes a Zaptel package of its own. 1.2.4 . Will they support a system that has unsupported kernel code? What is the alternative? buy support elsewhere. There are people who will support your Debian / Centos / whatever boxes. With RHEL and SuSE you have to buy support. With Debian it is optional. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
