[asterisk-users] Toshiba DK - Asterisk Integration
Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the Digium cards I should use in each of these cases and How should I integrate Asterisk with above systems. I read the article in http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and not sure whether that scenario fits mine. Also it is bit confusing to identify what Digium cards should I need for my cases. Any help is highly appreciated. Thanks, Indika. ___ --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] Stress-Testing Asterisk
On Tue, Nov 13, 2007 at 09:00:06AM +0100, Suity Zsolt wrote: Jeng Yu wrote: Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Try SIPp from HP (http://sipp.sourceforge.net/index.html) or SIP swiss army knife - SIPSAK (http://sipsak.org/) and also: Asterisk (http://asterisk.org/) is a highly programmable and configurable PBX. Use one on a stornger box or several client boxes. This can allow you to stress a box through SIP, IAX, H323, or whatever. (/me wonders if NBS could be used to stress-test an Asterisk box :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest official release you can expect a few troubles with Asterisk 1.4.11 (one way audio -- randomly, dropped calls). So you have to install the BETAS whether you want or not... That you have to use unique ports is a rumour and not SIP standard. As John said -- IP:Port must be unique . I definitely not understand why I should use random ports. Kind Regards, Erik I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. I use on every phone the 1 as local port and in the rtp.conf From my knowledge of IP I don't think this is a problem since the address/port would be unique. However the example config I originally had from Grandstream indicated that each phone should use a different port and recommended to use the random port option on the phones. I have since assigned the port number on each phone to 1 plus the extension number. This was done to create a unique port number and to help with troubleshooting when using Wireshark or tcpdump. I set this in the config file for each phone. John ___ --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] Record() : How to get filename created with %d?
In article [EMAIL PROTECTED],
Vincent [EMAIL PROTECTED] wrote:
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and you uwill see that the generated filename is available by using
${RECORDED_FILE}
Thanks for the tip. The reason I was looking for another solution is
that I couldn't get the value of the variable... but it's working now.
I'm not comfortable yet with functions/applications, and have no idea
what I did wrong :-/
BTW, what's the difference between functions and applications?
An application is a command executed by a dialplan priority, such as
Record, Verbose, TrySystem, etc. in your example below.
A function needs to be evaluated inside ${ } and returns a string value
that is substitued in place of the ${ }, such as STRFTIME in your
second example.
For those interested, here's some working code:
==
exten = 555,1,Record(/tmp/msg%d.wav,3,30)
exten = 555,n,Verbose(${RECORDED_FILE})
exten = 555,n,TrySystem(mv ${RECORDED_FILE}.wav /var/www/asterisk/)
The above line might be dangerous. The way %d works is that Record
tries %d=1, then 2 and so on, until it finds a value that doesn't
already exist.
If you have, say, foo%d as the filename, it creates foo1.wav, but then
when you move foo1.wav to a different directory, Record will use foo1.wav
again the next time.
You can avoid that by using ln instead of mv, but when you want to
delete the file, you must remember to do so in both places.
Or you could symlink one directory to another.
exten = 555,n,ExecIf($[${SYSTEMSTATUS} != SUCCESS],Verbose,Failed
moving WAV file)
==
Another way to generate a filename dynamically, using the current date
+ time:
==
exten =
_[1-4],n,Set(CALLTIME=${STRFTIME(${EPOCH},GMT+1,%d-%b-%Y-%Hh%M)})
exten = _[1-4],n,Record(/tmp/${CALLTIME}.wav,3,30)
exten = _[1-4],n,TrySystem(mv /tmp/${CALLTIME}.wav
/var/www/asterisk/)
exten = _[1-4],n,ExecIf($[ ${SYSTEMSTATUS} != SUCCESS
],Verbose,Failed moving WAV file)
==
That certainly would avoid the problem of reusing the same %d, but would
break if you had two calls in the same second.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Stress-Testing Asterisk
On 11/13/07, Jeng Yu [EMAIL PROTECTED] wrote: Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Hi, i don't want to make public announcments yet, but i'm preparing a open-source release of stress-test framework based on asterisk and php. There are few docs left, and i need final approvement from company's owner.. But - it should be ready within next few days.. So - take a look at sipp, etc, but afaik - there is no match to real-life call load simulation. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_alsa issue
Hi folks, Its the forth day I'm sticking to a problem with chan_alsa, The sound played or captured from the device is choppy time to time. I mean when talking on a console/dsp microphone the other side hear my sound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i hear the same choppy music on hold. Any one have any idea how i could get closer to the problem. Any hint would be highly appreciated. -- M. Shokuie Nia. _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --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] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info and old firmware versions at this link: http://www.grandstream.com/BETATEST http://www.grandstreamsucks.com Some feedbacks about firmware versions here: http://www.trixbox.org/forums/grandstream In the RTP port configuration I'm using the fixed 5004 and I think that you have to change it or use a random port only on some specific firewall/network topologies. My Asterisk version is the 1.4.12. Thank you and bye. Marco Signorini Original Message Subject: [asterisk-users] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 From:Erik Wartusch [EMAIL PROTECTED] Date:Tue, November 13, 2007 10:25 am To: asterisk-users@lists.digium.com -- Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest official release you can expect a few troubles with Asterisk 1.4.11 (one way audio -- randomly, dropped calls). So you have to install the BETAS whether you want or not... That you have to use unique ports is a rumour and not SIP standard. As John said -- IP:Port must be unique . I definitely not understand why I should use random ports. Kind Regards, Erik ___ --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] Default mohmp3 : free of rights ?
Hello Asterisk's Users ! Is anybody knows if the default MP3 tracks provided with the lastest release of asterisk is free of rights or not ? The default tracks are : - fpm-calm-river.mp3 - fpm-sunshine.mp3 - fpm-world-mix.mp3 Regards, -- Sébastien Mortier AbsysTech Tel : +33 892 460 991 Fax : +33 320 745 005 Gsm : +33 620 792 429 Assistante : Sarah Foucart [EMAIL PROTECTED] - http://www.absystech.fr Visitez le programme d'incentive AbsysTech : http://incentive.absystech.fr ___ --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] Default mohmp3 : free of rights ?
Sébastien Mortier wrote: Hello Asterisk's Users ! Is anybody knows if the default MP3 tracks provided with the lastest release of asterisk is free of rights or not ? From the doc directory: cat musiconhold-fpm.txt About Hold Music Digium has licensed the music included with the Asterisk distribution From FreePlayMusic for use and distribution with Asterisk. It is licensed ONLY for use as hold music within an Asterisk based PBX. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail hangup
Probably I was not able to explain myself properly however, for some measge this what happen -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' I cannot listen the message and the voicemailmain exists I am using asterisk 1.4.13 ___ --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] VoiceMail hangup
Il Neofita wrote: -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' It may be related to this bug: http://bugs.digium.com/view.php?id=11083 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [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] ACD Queue LOG RINGNOANSWER Content 0
Looks like a reject, as it continues to hunt to the next agent. I think it is a busy. Although I thought the queue shouldn't try an agent that is in use. Thanks BJ Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Monday, November 12, 2007 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0 Yes. That's supposed to to be the timeout value. In the case where it's 0 are you seeing a call reject or something else? asterisk wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? Thanks Doug ___ --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 -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ --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] Cisco 7911/7941/7970/7971 Softkey XML Files
Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn’t find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Toshiba DK - Asterisk Integration
Indika, The question of interface depends on how your Strata PBX are connected to the telco currently and what interfaces your Strata supports. If all you have is POTS interfaces to the telco, your integration may be limited because every SIP extension will require a separate POTS line to the Strata. But if you have a T1 interface, you should be able to have trunked lines/multiple extensions. So we need more details. Tony Plack Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the Digium cards I should use in each of these cases and How should I integrate Asterisk with above systems. I read the article in http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and not sure whether that scenario fits mine. Also it is bit confusing to identify what Digium cards should I need for my cases. Any help is highly appreciated. Thanks, Indika. ___ --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] [Fwd: Re: VoiceMail hangup]
Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: You have 1 new message. Press 1 for new messages, press 2 for... or # to exit (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message. But, if: - I dial the voicemail extension. - I hear you have 1 new message. Press 1... 1 pressed (without waiting for the message playing) - Asterisk hangups. I'm not always able to replicate the problem but, as Il Neofita, I'm using the italian prompts... could be a problem related to that? Bye and regards Marco Signorini. Il Neofita wrote: -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' It may be related to this bug: http://bugs.digium.com/view.php?id=11083 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to play Asterisk .raw file
I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary ___ --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] [Fwd: Re: VoiceMail hangup]
Hi I have the same problem On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote: Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: You have 1 new message. Press 1 for new messages, press 2 for... or # to exit (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message. But, if: - I dial the voicemail extension. - I hear you have 1 new message. Press 1... 1 pressed (without waiting for the message playing) - Asterisk hangups. I'm not always able to replicate the problem but, as Il Neofita, I'm using the italian prompts... could be a problem related to that? Bye and regards Marco Signorini. Il Neofita wrote: -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' It may be related to this bug: http://bugs.digium.com/view.php?id=11083 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Conference rooms
I all, I have a question about the use of conference rooms: can I, with a Voip telephone or softphone call some other telephone and invite them in a conference room? I read a lot of documentations about asterisk, but i can't find any example ! Thanks, best regard Fabio ___ --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] Record() : How to get filename created with %d?
On Tue, 2007-11-13 at 01:27 +0100, Vincent wrote:
BTW, what's the difference between functions and applications?
Functions are pretty much like applications, but the difference is:
Functions can return a value, and you can use them to set a value as
well, depending on the function
Applications do not really return a value (besides a result code that
copuld terminate the dialplan)
For instance, you can say Set(x=${CDR(userfield)} to set x to the value
of the userfield variable in the CDR, and you can also say
Set(CDR(userfield)=${x}) to set that field to the value of x. DB, CDR,
and others can be used to set/get various values.
You can read the doc/channelvariables document, and also the 'core show
applications', 'core show functions' and 'core show application xxx' and
'core show function XXX' to get documentation on every app and func that
is installed in asterisk.
murf
--
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Software Developer
Digium
smime.p7s
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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
There is an option to specify a softkey file in SEPmac.cnf.xml. I have an email into our Cisco rep. I'm hoping he can shed some light on this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, November 13, 2007 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] ACD Queue LOG RINGNOANSWER Content 0
On 11/12/07, asterisk [EMAIL PROTECTED] wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? For my internal reporting, I consider 0 or 1000 to be the result of a phone being on DND. Since all the time values in the log are rounded to the nearest 1000, I speculate that 0 is a rejection in 500ms and 1000 is a rejection in the 500 to 1499 ms range. I figure unless someone is hovering over the ignore button on a softphone, they aren't going to be able to click it so fast that Asterisk registers it as RINGNOANSWER|1000. Likewise, RINGNOANSWER|2 is (for me, given timeout=20 in queues.conf) a failure to pick up a presented call. Everything from 2000 through 19000 I treat as a manual ignore triggered by the agent. So far, the reports I generate based on these rules seem to make sense to the managers reading them. -- j. ___ --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 to integrate Asterisk with Avaya
Hello Everyone, Can someone please point to sources how to integrate Asterisk PBX with Avaya..? What normalize and expose protocol/API does Avaya support which can be use with Asterisk? Thanks in advance, -C ___ --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] Conference rooms
Hi Fabio, Once you have an Asterisk box that have a conference room configured and a VoIP phone the supports forward you can easily forward your guests to the conference room. Moreover you can create a conference room extension available, via password, from the PSTN line. Hope this can help you. On Nov 13, 2007 3:38 PM, Fabio Cappelletti [EMAIL PROTECTED] wrote: I all, I have a question about the use of conference rooms: can I, with a Voip telephone or softphone call some other telephone and invite them in a conference room? I read a lot of documentations about asterisk, but i can't find any example ! Thanks, best regard Fabio ___ --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] Install Scripts for CentOS 4
Hi all, I had a few people ask me for the install scripts that I created for CentOS 4. OK, here they are. The fist link install asterisk-freepbx-spandsp-nv_faxdetect-tx/rxfax_app. The second on install hylafax-iaxmodem. http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh http://jonnt.users.taylortelephone.com/asterisk/iax-hylafax-setup.sh Have fun and hope this helps. Jonn ___ --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] [Fwd: Re: VoiceMail hangup]
Am Dienstag, den 13.11.2007, 09:29 -0500 schrieb Il Neofita:
Hi
I have the same problem
On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote:
Hi Neofita, Doug and All.
I think I've the same problem but I don't know if it's related to the bug
suggested below.
I try to explain my behavior:
- I dial the voicemail extension.
- I hear: You have 1 new message. Press 1 for new messages, press 2 for...
or # to exit (I listen the complete message or most part of it)
- I press 1
- I can hear the first recorded message.
But, if:
- I dial the voicemail extension.
- I hear you have 1 new message. Press 1... 1 pressed (without waiting
for the message playing)
- Asterisk hangups.
I'm not always able to replicate the problem but, as Il Neofita, I'm
using the italian prompts... could be a problem related to that?
Bye and regards
Marco Signorini.
Marco, Il Neofita,
it seems you exactly found that bug. May I suggest a workaround idea:
After the dialplan call to VoiceMail() for leaving the message, call an
AGI script that checks the related directory, especially the last
message. If it is less than 45 bytes, remove it. That AGI need not be
too complicated, might be a bash script like
#!/bin/bash
for A in /var/spool/asterix/voicemail/default/* ; do
for B in ${A}/*.wav ; do
SIZE=`ls -l --color=never ${B} | awk {print \$5; }`
if [[ $SIZE -le 44 ]] ; then
rm -f $B
fi
done
done
Caveat emptor, just a quick idea.
Trying another file format for voicemail recording might also be an
option, as this seems to relate to the wav header somehow. You might
choose alaw, ulaw, perhaps gsm or speex... Give it a try, and report
back if that helps. The voip-info.org wiki page about voicemail.conf
should tell you how to exactly set that up.
BR
Anselm
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Re: [asterisk-users] How to play Asterisk .raw file
Gary wrote: I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary That's SLINEAR. I know CoolEdit, now Adobe Audition can play them. Not sure about Audacity. I've never tried it with that. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] How to play Asterisk .raw file
Gary wrote: I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary I believe .raw files are slinear (signed linear). They are effectively wav files without a header. You can use sox to convert them to your preferred format. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to play Asterisk .raw file
Gary wrote: I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? I don't know, but you can import a raw audio file into audacity making different parameter selections, eg. sampling rate ( 8khz ) and format ( ulaw, gsm ) -- ___ --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] RTP traffic not being forwarded
On Tue, 2007-11-13 at 15:26 +1100, Ryan Newington wrote: Hi Vivek, Thanks for the link. I had a look through and couldn’t find anything that worked. There are no NAT problems as this is all taking place on my internal network. The rtp.conf is used to configure the ports. There are no firewalls or gateways in between these devices. Asterisk is listening on the correct ports, and receiving the traffic, as no ICMP messages are being generated to say that the packets could not be delivered. I once had this problem and it was because I had not set localnet in sip.conf to the proper network. When I fixed it I got audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Scripts for CentOS 4
Hi Jonn will these scripts work with CentOS 5? --Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Tuesday, November 13, 2007 10:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Install Scripts for CentOS 4 Hi all, I had a few people ask me for the install scripts that I created for CentOS 4. OK, here they are. The fist link install asterisk-freepbx-spandsp-nv_faxdetect-tx/rxfax_app. The second on install hylafax-iaxmodem. http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh http://jonnt.users.taylortelephone.com/asterisk/iax-hylafax-setup.sh Have fun and hope this helps. Jonn ___ --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] ACD Queue LOG RINGNOANSWER Content 0
Cool thanks James... Doug From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Tuesday, November 13, 2007 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD Queue LOG RINGNOANSWER Content 0 On 11/12/07, asterisk [EMAIL PROTECTED] wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? For my internal reporting, I consider 0 or 1000 to be the result of a phone being on DND. Since all the time values in the log are rounded to the nearest 1000, I speculate that 0 is a rejection in 500ms and 1000 is a rejection in the 500 to 1499 ms range. I figure unless someone is hovering over the ignore button on a softphone, they aren't going to be able to click it so fast that Asterisk registers it as RINGNOANSWER|1000. Likewise, RINGNOANSWER|2 is (for me, given timeout=20 in queues.conf) a failure to pick up a presented call. Everything from 2000 through 19000 I treat as a manual ignore triggered by the agent. So far, the reports I generate based on these rules seem to make sense to the managers reading them. -- j. ___ --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] sangoma zaptel patches
Dovid B wrote: - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 11, 2007 8:21 PM Subject: Re: [asterisk-users] sangoma zaptel patches On Sunday 11 November 2007 11:07:04 Steve Totaro wrote: Tzafrir Cohen wrote: Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with that patch. It seems that under certain circumstances it may cause unexpected behavior when used with other Zaptel channel drivers. I also don't understand why a safer method is not used. Just out of curiosity, I have yet to see any issues with Sangoma cards and the way they ride on top (and patch) the Zaptel drivers. This personal dataset is around one hundred productions boxes. How many of those boxes are of the type that Tzafrir is worried about? Specifically, how many of those boxes contain a combination of telephony hardware from vendors other than Sangoma? I have a box that now has a TDM400P. I will be installing a sangoma card in it soon and I actually need support for this. I set up almost the exact same configuration and all went well (HP DL380). No gotchas or glitches. I have a feeling that Tzafrir is trying to fix what is not broken, since he never pointed out a single conflict between various hardware using patched Zaptel drivers configurations. Maybe he is looking down the road and being proactive which I applaud, but I think he is obsessing over what he feels is the incorrect way of doing things and demanding (tone in emails) that they cooperate and do what he tells them. A little tact goes a long way. Thanks, Steve ___ --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] route INVITE sip:[EMAIL PROTECTED]
Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an inbound route! It matches a DID number. How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U 217.36.112.145:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: [EMAIL PROTECTED] Contact: sip:217.66.118.145:5060. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: 0614740696 sip:[EMAIL PROTECTED];user=phone;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: sip:[EMAIL PROTECTED];user=phone. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 13 Nov 2007 18:07:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 233. . ___ --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] Install Scripts for CentOS 4
No, I am working on that. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 13, 2007 10:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Install Scripts for CentOS 4 Hi Jonn will these scripts work with CentOS 5? --Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Tuesday, November 13, 2007 10:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Install Scripts for CentOS 4 Hi all, I had a few people ask me for the install scripts that I created for CentOS 4. OK, here they are. The fist link install asterisk-freepbx-spandsp-nv_faxdetect-tx/rxfax_app. The second on install hylafax-iaxmodem. http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh http://jonnt.users.taylortelephone.com/asterisk/iax-hylafax-setup.sh Have fun and hope this helps. Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Scripts for CentOS 4
OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? --Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Tuesday, November 13, 2007 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Install Scripts for CentOS 4 No, I am working on that. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 13, 2007 10:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Install Scripts for CentOS 4 Hi Jonn will these scripts work with CentOS 5? --Zaheer ___ --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] Install Scripts for CentOS 4
Older base packages (older MySQL, etc). As far as overall running Asterisk, you're not liable to run into anything negative on the 4.5 side as opposed to 5. N. Zaheer K. Master wrote: OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? --Zaheer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Tuesday, November 13, 2007 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Install Scripts for CentOS 4 No, I am working on that. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 13, 2007 10:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Install Scripts for CentOS 4 Hi Jonn will these scripts work with CentOS 5? --Zaheer ___ --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] Install Scripts for CentOS 4
On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote: OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? some of the packages in 4 (4.5) are pretty old, eg. sox that is bundled with 4.5 does not recognize several common audio formats. You can upgrade some of the packages to more current versions. If you plan to do any asterisk-AGI using php, centos 4.5 installs php4, most of the new php stuff is written in php5. I would also suggest evaluating debian as an alternative to centos 4 5 for your * box. In my limited experience, I have found debian 4.0 (Etch) to be a better platform for * than centos 4.5 5. But please, I don't want to start a distro flame war here. YMMV. -baji. -- ___ --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] Install Scripts for CentOS 4
On Tue, 2007-11-13 at 14:13 -0500, Zaheer K. Master wrote: OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? I heard that the soon available CentOS 5.1 will have high resolution timer support in its kernel. If you use only ztdummy this improves timing quite a bit. For that reason alone I would prefer CentOS 5.1. Regards, Patrick ___ --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] sangoma zaptel patches
Steve Totaro wrote: Dovid B wrote: - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 11, 2007 8:21 PM Subject: Re: [asterisk-users] sangoma zaptel patches On Sunday 11 November 2007 11:07:04 Steve Totaro wrote: Tzafrir Cohen wrote: Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with that patch. It seems that under certain circumstances it may cause unexpected behavior when used with other Zaptel channel drivers. I also don't understand why a safer method is not used. Just out of curiosity, I have yet to see any issues with Sangoma cards and the way they ride on top (and patch) the Zaptel drivers. This personal dataset is around one hundred productions boxes. How many of those boxes are of the type that Tzafrir is worried about? Specifically, how many of those boxes contain a combination of telephony hardware from vendors other than Sangoma? I have a box that now has a TDM400P. I will be installing a sangoma card in it soon and I actually need support for this. I set up almost the exact same configuration and all went well (HP DL380). No gotchas or glitches. I have a feeling that Tzafrir is trying to fix what is not broken, since he never pointed out a single conflict between various hardware using patched Zaptel drivers configurations. Maybe he is looking down the road and being proactive which I applaud, but I think he is obsessing over what he feels is the incorrect way of doing things and demanding (tone in emails) that they cooperate and do what he tells them. A little tact goes a long way. I think that part of it is that the patch that they do to zaptel replicates existing zaptel functionality (zt_hdlc functions) for hardware d-channel support. There has been no change in their patch to use these existing functions, and they are implementing this via an ioctl function within a kernel driver, which is not a pretty way to do what they are trying to do. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Scripts for CentOS 4
Baji Panchumarti [EMAIL PROTECTED] writes: On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote: OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? some of the packages in 4 (4.5) are pretty old, eg. sox that is bundled with 4.5 does not recognize several common audio formats. You can upgrade some of the packages to more current versions. If you plan to do any asterisk-AGI using php, centos 4.5 installs php4, most of the new php stuff is written in php5. I would also suggest evaluating debian as an alternative to centos 4 5 for your * box. In my limited experience, I have found debian 4.0 (Etch) to be a better platform for * than centos 4.5 5. But please, I don't want to start a distro flame war here. YMMV. The nice thing about CentOS (as opposed to Redhat proper) is that they provide the CentOS Plus repository, so installing PHP5/MySQL would be something like: # yum --enablerepo=centosplus install php php-mysql -- Kyle Sexton ___ --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] Install Scripts for CentOS 4
Except that FreePBX has problems with php5. I played with this a little with asterisk 1.4 and CentOS 5 and got it all working, I just have not had time to get the scripts finised. The other thing with CentOS vs Debian is that CentOS packages do not change every month or so. Debain seems to a little more on the bleeding edge of this, which is not the best thing for a production system. It is totally person preference and nothing else. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Sexton Sent: Tuesday, November 13, 2007 2:30 PM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Install Scripts for CentOS 4 Baji Panchumarti [EMAIL PROTECTED] writes: On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote: OK Thanks! If I'm building a new Asterisk system from scratch, is there any downside to using CentOS 4 instead of 5? some of the packages in 4 (4.5) are pretty old, eg. sox that is bundled with 4.5 does not recognize several common audio formats. You can upgrade some of the packages to more current versions. If you plan to do any asterisk-AGI using php, centos 4.5 installs php4, most of the new php stuff is written in php5. I would also suggest evaluating debian as an alternative to centos 4 5 for your * box. In my limited experience, I have found debian 4.0 (Etch) to be a better platform for * than centos 4.5 5. But please, I don't want to start a distro flame war here. YMMV. The nice thing about CentOS (as opposed to Redhat proper) is that they provide the CentOS Plus repository, so installing PHP5/MySQL would be something like: # yum --enablerepo=centosplus install php php-mysql -- Kyle Sexton ___ --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] Call Forward on SIP unreachable (network failure)
Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on unavailable (ring but no answer) not nothing about CF on (SIP) unreachable. Thank you and best regards, Antoine Megalla. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install Scripts for CentOS 4
On Tue, Nov 13, 2007 at 02:45:38PM -0600, Jonn R Taylor wrote: The other thing with CentOS vs Debian is that CentOS packages do not change every month or so. Debain seems to a little more on the bleeding edge of this, which is not the best thing for a production system. It is totally person preference and nothing else. What Debian? Debian has the Stable distribution that has the same policy as RHEL/CentOS: critical bugfixes only (security updates are such bug fixes) with every attempt made to minimize impact on system. Now if you're using a non-stable Debian version for production, this stability is never guaranteed. -- 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] Install Scripts for CentOS 4
On Nov 13, 2007 4:30 PM, Kyle Sexton wrote: [...] The nice thing about CentOS (as opposed to Redhat proper) is that they provide the CentOS Plus repository, so installing PHP5/MySQL would be something like: # yum --enablerepo=centosplus install php php-mysql Amen ! I have used centosplus and found it to be a very valuable companion resource for centos. On debian the equivalent is aptitude install php5-common php5-cli php5-dev php5-mysql one difference I found between using yum on centos aptitude on debian, and I am not sure if it is universally the case, is that the packages hit the ground running with debian. Eg. I run aptitude install ntp and the system clock syncs in a few secs, I run aptitude install mysql-server mysql-client and mysql server is running. whatever happened to the need to dig around in obscure directories looking in obscure .conf files, after installing a package, trying to decide if you can have space between the option -p the value :-) -baji. -- ___ --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] ztdummy, zttest
Well, I got it working. Come to find out that it looks like version 2.6.18 of the kernel has issues with the RTC. It has little to do with any of the other things that I tried. I upgraded the kernel to 2.6.23.1 and on the first try, it comes up and runs with TSC. ztdummy is working for me now. I used the oldconfig to update the kernel from the debian 2.6.18-5-686 and compiled. Runs well. Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than other machine that is working. CPU is a HP, Pentium 4 (same than other machine), even I loaded the same bristuff than that machine (who doesn't have any specific hardware, now). I couln't make zttest (well, ztdummy) to run. I tried different versions of bristuff+asterisk, I also tried to load and not load zaptel, qozap. Nothing. I got an rtc "Warning" message, something like "some interruptions has been lost at 1024Hz" (aprox.). Any clue where to check? USB modules are the same than other machine... We was completely confused about it (how to fix, I mean). 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] Install Scripts for CentOS 4
funny, I thought Gentoo was on the bleeding edge, and Debian was behind CentOS guess it is a matter of opinion. On Tue, Nov 13, 2007 at 02:45:38PM -0600, Jonn R Taylor wrote: The other thing with CentOS vs Debian is that CentOS packages do not change every month or so. Debain seems to a little more on the bleeding edge of this, which is not the best thing for a production system. It is totally person preference and nothing else. What Debian? Debian has the Stable distribution that has the same policy as RHEL/CentOS: critical bugfixes only (security updates are such bug fixes) with every attempt made to minimize impact on system. Now if you're using a non-stable Debian version for production, this stability is never guaranteed. ___ --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] Install Scripts for CentOS 4
On Nov 13, 2007 Jonn R Taylor wrote: [...] The other thing with CentOS vs Debian is that CentOS packages do not change every month or so. Debain seems to a little more on the bleeding edge of this, which is not the best thing for a production system. It is totally person preference and nothing else. funny you should say that today, because a short while before your post, one of the open source enthusiasts in my neck of the woods was complaining for exactly the opposite reasons, see below :-) -- Mat wrote on Nov 13th -- That is one of my biggest problems with Debian, by the time a new release is out you are already behind the curve. With it's insanely slow release path, you are forced to look towards other repositories or mix testing/unstable with stable. Gets a bit out of hand. you are right, it is definitely subjective to ones perspective. -baji. -- ___ --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] sangoma zaptel patches
Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with that patch. It seems that under certain circumstances it may cause unexpected behavior when used with other Zaptel channel drivers. I also don't understand why a safer method is not use The only beef I have about Sangoma drivers is that on one ocassion, an upgrade to one of the 'stable' releases caused a reproducible kernel panic. It was so bad the box would panic in mid-boot. If this was a remote upgrade I would have been in big trouble. Luckily I was on-site and Sangoma support was quick and provided a fix within 24 hours. But something like this has never happened with any Digium cards I have worked with. This incident has caused me to have second thoughts about installing Sangoma cards at datacenters and remote locations. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
Could you describe in detail how did you fall into this situation, I mean the real example which SIP phone sends this invite? Is registered in asterisk? it is a non-registered sip phone trying to dial a sip user at your * box? If this is an issue with a specific hardware outside of your asterisk, may be something not well configured ... describe it a bit more in detail. If you don't have anyworkaround for this Invite format I would use OpenSER in front of Asterisk to handle this invites and replace to SIP URI with info from the tag TO: ... Any way if you provide more details may be someone in the Mailing list is able to help u out;) Best regards MoutaPT On Nov 13, 2007 6:14 PM, Marc LEURENT [EMAIL PROTECTED] wrote: Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an inbound route! It matches a DID number. How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U 217.36.112.145:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: [EMAIL PROTECTED] Contact: sip:217.66.118.145:5060. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: 0614740696 sip:[EMAIL PROTECTED];user=phone;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: sip:[EMAIL PROTECTED];user=phone. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 13 Nov 2007 18:07:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 233. . ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forward on SIP unreachable (network failure)
${DIALSTATUS} will be one of:
- *CHANUNAVAIL* : Channel unavailable (for example in sip.conf, when
using qualify=, the SIP chan is unavailable)
- *BUSY* : Returned busy
- *NOANSWER* : No Answer (i.e SIP 480 or 604 response)
- *ANSWER* : Call was answered
- *CANCEL* : Call attempt cancelled (i.e user hung up before the call
connected)
- *DONTCALL* : Privacy manager don't call
- *TORTURE* : Privacy manager torture menu
- *CONGESTION* : Means Congestion, or anything else (some other error
setting up the call)
Did you test CHANUNAVAIL or CONGESTION ?
Debug DIALSTATUS var for this case using Noop application in dialplan.
On Nov 13, 2007 8:51 PM, Antoine Megalla [EMAIL PROTECTED] wrote:
Hi,
I am trying to implement call forwarding on the event
that my ATA was not
reachable to Asterisk, whether due to registration
timeout, network
interruptions between the ATA and Asterisk, or simply
because the network on
which the ATA resides in unreachable for any reason.
I there a way of implementing such a feature in
Asterisk?
I have implemented CF unconditional, and CF on busy,
CF on unavailable (ring
but no answer) not nothing about CF on (SIP)
unreachable.
Thank you and best regards,
Antoine Megalla.
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
as far as I know, softkey layout is managed by Cisco Call Manager and only available running on skinny protocol. On Nov 13, 2007 2:50 PM, Anciso, Roy [EMAIL PROTECTED] wrote: There is an option to specify a softkey file in SEPmac.cnf.xml. I have an email into our Cisco rep. I'm hoping he can shed some light on this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, November 13, 2007 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] function voicemailmain
Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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] Nortel digital FXO channel bank? Exists?
We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD ___ --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 to pay for libpri development
Can someone advise on how to go about finding someone QUALIFIED to make changes to libpri? We have a pilot stuck on hold, due to old buggy PRI software on a meridian PBX. Upgrading the meridian software is not an option, sowe would like to have libpri changed to compensate for the bug. Is this a digium only type fix? I've called digium support but they only offer support for their hardware - they said they can't help with software fixes (even though we are happy to pay). I already tried paying the asteriskguru web site guys for tech support, but after $300 all they have done is confirm there is a software bug on the meridian. Wow, that was a waste of money I don't want to throw too much more money down a black hole. Can someone suggest where to turn for this? Thanks, MD ** I thought of posting on the commercial asterisk list, but I'm afraid of every unqualified developer jumping up for the money. Hopefully the user community can comment first. ** ___ --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] Nortel digital FXO channel bank? Exists?
Quoting Michelle Dupuis [EMAIL PROTECTED]: We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? I think the basic wireline signalling is isdn bri for that, but with non-standard protocols. ie the channel bank would talk to the ksu/pbx, but nothing but a reverse of the same hardware would understand anything on the other end of the t1. ps - this sort of channel bank would be pricey at best, rediculous probably in reality. I would try other solutions before even attempting this sort of thing since the odds of success are probably not too high. Thanks MD Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to pay for libpri development
Quoting Michelle Dupuis [EMAIL PROTECTED]: agree with whoever you choose what the price is, but make most of it payable on delivery of working code, that will separate those that can actually do it from those who can't or aren't sure. even more fair to both parties put the money in escrow so there is no risk to either party for a well defined deliverable. I am sure there are many people on the list that are up to the task. Can someone advise on how to go about finding someone QUALIFIED to make changes to libpri? We have a pilot stuck on hold, due to old buggy PRI software on a meridian PBX. Upgrading the meridian software is not an option, sowe would like to have libpri changed to compensate for the bug. Is this a digium only type fix? I've called digium support but they only offer support for their hardware - they said they can't help with software fixes (even though we are happy to pay). I already tried paying the asteriskguru web site guys for tech support, but after $300 all they have done is confirm there is a software bug on the meridian. Wow, that was a waste of money I don't want to throw too much more money down a black hole. Can someone suggest where to turn for this? Thanks, MD ** I thought of posting on the commercial asterisk list, but I'm afraid of every unqualified developer jumping up for the money. Hopefully the user community can comment first. ** Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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-users Digest, Vol 40, Issue 37
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[asterisk-users] OT: Re: How to pay for libpri development
TP'n to follow flow. Seems lately (for me at least), if i did the pay on completion, i was the one that got screwed over. I obviously do not do that anymore. Sometimes you have to change your methods regardless of your abilities. Jon Pounder wrote: Quoting Michelle Dupuis [EMAIL PROTECTED]: agree with whoever you choose what the price is, but make most of it payable on delivery of working code, that will separate those that can actually do it from those who can't or aren't sure. even more fair to both parties put the money in escrow so there is no risk to either party for a well defined deliverable. I am sure there are many people on the list that are up to the task. *snipped ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is wrong with this mailing list
Hi all, Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? Regards. _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --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] function voicemailmain
On Tuesday 13 November 2007 21:34:31 Rilawich Ango wrote: Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? No. -- 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] What is wrong with this mailing list
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote: Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit Reply to an existing message and just change the subject line. -erik ___ --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] MOH Codec Issue - Fixed
I recompiled a version of Zaptel to no avail in an attempt to find a quick fix. This did not help, however have since upgraded the box to Asterisk 1.4.13 and the issue has disappeared. As such I put it down to either being; 1. Zaptel was broken, I should have however recompiled Asterisk after recompiling Zap (Opposed to being impatient and frustrated), or 2. There is a bug in 1.4.7. I haven't had time to try and reproduce it, plus it would be a purely academic project as if there was a bug it has since been fixed. Thanks for the suggestions Paul. Nick. On 13/11/07 4:48 PM, Paul Hales wrote: Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC connection to Microsoft SQL Server
On Monday 12 November 2007 18:14:57 Robert McNaught wrote: I wish to integrate a Microsoft SQL server with Asterisk for CDRs and for dialplan routing based on database values, and have this application scale to a large number of simultaneous calls: The Asterisk: The Future of Telephony 2nd edition book states that: ‡ The pooling and limit options are quite useful for MS SQL Server and Sybase databases. These permit you to establish multiple connections (up to limit connections) to a database while ensuring that each connection has only one statement executing at once (this is due to a limitation in the protocol used by these database servers). Does this suggest any kind of performance issue with scaling? If there is any such issue, it is due to a limitation with the database server protocol, not with Asterisk. The description should have made that clear. I am assuming not as all this indicates is that DB queries are pooled from the ODBC connection on the Asterisk Server side rather than the SQL Server? MS SQL Server is simply unable to accept multiple simultaneous queries on the same connection, so this is the only solution possible to avoid any collisions. Has anyone done this before in a large implementation? I've never done it in a large implementation, but I don't see any reason why it wouldn't work. If you're really concerned with the scale of opening a lot of connections, might I suggest that you use PostgreSQL? -- 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] What is wrong with this mailing list
On Nov 14, 2007 12:21 AM, Mohammad Shokuie wrote: Hi all, Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? Regards. not this time, came thru fine. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! Regards. -- M. Shokuie Nia Date: Tue, 13 Nov 2007 23:33:08 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What is wrong with this mailing list On Nov 13, 2007 11:21 PM, Mohammad Shokuie wrote: Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit Reply to an existing message and just change the subject line. -erik ___ --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 _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
On Nov 13, 2007 11:44 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote: HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! You're using firefox? How so? I'd recommend either a good email client (Thunderbird) or a good web email interface (gmail). (I'm using gmail's web interface) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
Erik Anderson wrote: On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote: Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit Reply to an existing message and just change the subject line. Which will screw up any threading people might be using in their e-mail program, as well as screw up the threading in the list archives. Doing this is an act of desperation and should be discouraged. Generally people that experience this problem either have overly aggressive spam filters or they are sending from an address different from the one the subscribed from. ___ --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] function voicemailmain
vi app_voicemail.c On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --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] What is wrong with this mailing list
Hi Erik, By firefox i mean a Hotmail web mail, it means there is no mail client. I dont know if there would be any difference if i subscribe and use other mails like gmail! Regards. -- M. Shokuie Nia Date: Tue, 13 Nov 2007 23:52:03 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What is wrong with this mailing list On Nov 13, 2007 11:44 PM, Mohammad Shokuie wrote: HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! You're using firefox? How so? I'd recommend either a good email client (Thunderbird) or a good web email interface (gmail). (I'm using gmail's web interface) ___ --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 _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx ___ --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] MOH Codec Issue - Fixed
I have had Asterisk play up very badly when Zaptel is not running quite right (or misconfigured) - no audio at all. PaulH On Wed, 2007-11-14 at 16:17 +1100, Nick Brown wrote: I recompiled a version of Zaptel to no avail in an attempt to find a quick fix. This did not help, however have since upgraded the box to Asterisk 1.4.13 and the issue has disappeared. As such I put it down to either being; 1. Zaptel was broken, I should have however recompiled Asterisk after recompiling Zap (Opposed to being impatient and frustrated), or 2. There is a bug in 1.4.7. I haven't had time to try and reproduce it, plus it would be a purely academic project as if there was a bug it has since been fixed. Thanks for the suggestions Paul. Nick. On 13/11/07 4:48 PM, Paul Hales wrote: Is it possibly a funny zaptel issue? Paul Hales AsteriskIT On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote: Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/10.97.1.33-09f0cfc8, sounds) in new stack [Nov 13 15:00:14] WARNING[5461]: channel.c:2964 set_format: Unable to find a codec translation path from alaw to unknown [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/10.97.1.33-09f0cfc8' to format 'unknown' -- Started music on hold, class '?S?', on channel 'SIP/10.97.1.33-09f0cfc8' [Nov 13 15:00:14] WARNING[5461]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'sounds') on channel SIP/10.97.1.33-09f0cfc8 Have attempted to use an alternate Music On Hold context and forced a format= within musiconhold.conf. Otherwise all other audio (Playback, voice etc) seems fine. Anyone seen this before? Can not see anything in the tracker regarding this issue in 1.4.7 specifically. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
On Nov 14, 2007 12:52 AM, Eric ManxPower Wieling wrote: [...] Generally people that experience this problem either have overly aggressive spam filters or they are sending from an address different from the one the subscribed from. he has a hotmail address, my money is on their bulk mail filter. How much you wanna bet, he'll find his posts in his own bulk mail folder. Mohammad, please add this address to your addr book asterisk-users@lists.digium.com so hotmail will always allow emails from that address. hth, -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk trunk and manager redirect problem
Dear All, Have anyone tested the trunk version and redirect command, it seems the pbx routines changed much and the redirect mechanism doesnt work well with this new changes. When ever i redirect a channel i got the channel hanged up. After a survey in the code i got that when the channel soft hanged up in the async goto the loop in the pbx_run exits and the channel got a real hang up instead of jumping to the begining of the loop in the routine. Regards. -- M. Shokuie Nia _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE ___ --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] function voicemailmain
You mean modify the source? Could you give me an example, say I wrong to remove advance option? On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: vi app_voicemail.c On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, Can I simply the voicemailmain IVR? I just only want some of the option in voicemailmain, ie read or delete messages. Is it possible to configure that function? Ango ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
