[asterisk-users] asterisk novice needs help.
The basic incoming and out going works fine. Trying to create a basic dial plan and asterisk hangs up on me. First issue: I am only hearing part of the recording when I call in Second issue; Before I am even able to choose an option of hear the rest of the recording, asterisk hangs up on me. Any assistance would truly be appreciated. Brad My dial plan of issues... ;exten = s,1,Answer() ;exten = s,2,DIAL(SIP/100,20) exten = s,1,Answer(60) exten = s,2,Background(otherwise-press) exten = s,1,Playback(digits/1) exten = s,2,Goto(default,s,1) exten = s,1,Playback(digits/2) exten = s,2,Goto(default,s,1) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(incoming,s,1) exten = t,1,Playback(vm-goodbye) ;exten = t,2,Hangup( ) ___ --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] Novice needs help part 2
PS, I have zero FX gear. I am 100% SIP Brad ___ --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] Novice needs help part 2
Welcome to the Asterisk Users mailing list, On Fri, Jul 20, 2007 at 02:11:08AM -0400, BSumrall wrote: PS, I have zero FX gear. I am 100% SIP Huh? This nmust be related somehow to the PRI Busy problem thread you've answered to, otherwise you wouldn't have replied to it, right? If you want to post a new message, start a new message. Don't just reply to an arbitrary list message and change the subject and contents. Check the list's archives and see that the threading has remained. If you want to follow-up on someone's message, then please reply to it. This will maintain threading. Thus I can easily go one message up and see what signature that PS refers to. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use call transfer
I have SNOM phone and in my phone there is a transfer button but whn i use transfer key and enter another party number i got hangup so is there any configuration for Dial() t, T option is there any need to specifiy t or T option in dial plan Andrew Joakimsen [EMAIL PROTECTED] wrote: On 7/19/07, satish patel wrote: I have snom SI 120 sip phone and there is transfer button but id there any configuration in asterisk part for call transfer feature ??? Nothing else is required. Since the phone has a transfer button there is no need to use features.conf. What happens is the call is placed just like a regular phone call and then the phone indicates to Asterisk how the call should be transfered. This is the normal behaviour of SIP and there is no configuration for it. Usually you press transfer, dial the number to transfer to and then press transfer again, but I've never used a SNOM phone so I wouldn't know how those work. ___ --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 a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem after upgrading from 1.2.21.1 to 1.2.22
You should be running the latest Zaptel LibPRI both of which recently have been updated. We run a similar configuration and have not seen this problem with the upgrade. Even after upgrading Zaptel to the latest (1.2.19) from 1.2.17.1 there is the same problem. Libpri-1.2.5 was already at the lastest. Asterisk-1.2.22 won't work but asterisk-1.2.21 works fine. I still get the following in '/var/log/asterisk/message' pbx.c: Cannot find extension '' in context '(null)' Again, nothing has changed in the configuration files. Going back to 1.2.21 corrects the problem. Outgoing calls work fine. Internal calling works fine. (Internal phones are some SIP phones and some Zap phones hanging off of a channel bank) The only lines I am having problems with are our DID trunks incoming from the phone company. They are on a channelized T1 (but not PRI) set up with signalling=em_w in zapata.conf. I have done 'make clean; make install' on zaptel libpri asterisk asterisk-addons Pretty much whenever a new release of asterisk has come out in the 1.2 branch I have updated. (sometimes zaptel gets a little further behind since there is a short outage to upgrade) I have never had any problems before so I believe I am being thorough with the upgrade procedures. Again, has anything changed in how the zapata.conf is parsed? What about extensions.conf when using #includes? I'm not sure what else to do. Does anyone have any other suggestions? Don Pobanz winmail.dat___ --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] awful list delays: 4 days!
Lenz (?), you are not the only one! It took about five days for Anthony's reply to reach me. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Friday, June 29, 2007 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] awful list delays: 4 days! Andres Paglayan wrote: On Jun 29, 2007, at 12:50 PM, Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. I'll let you know next week, ;^) -- Loway Research - Home of QueueMetrics http://queuemetrics.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 Andres Paglayan --Harmony is more important than being right Bapak ___ --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 ROFL, yeah its you. I see posts within a few hours. ___ --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] pattern base call routing
Dear all I have 2 E1 card on my asterisk and i want to route call with fix pattern like if anyone dial mobile number like 9818875535 so it will use PRI 1 and rest of the world goes through PRI 2 means whn number prefix 98XX then call goes through specified E1 is it possible ??? satish patel - Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. ___ --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] * core file not recognized
On Thu, Jul 19, 2007 at 08:32:35PM -0700, Jay Wilton wrote: Hello, I am trying to debug a machine that segfaults. A core file is produced like /tmp/core.4545 . The command and error: gdb /usr/sbin/asterisk -c /tmp/core.4545 GNU gdb 6.3-debian ...CUT This GDB was configured as i386-linux...Using host libthread_db library /lib/libthread_db.so.1. /tmp/core.4545 is not a core dump: File format not recognized So what is that file? file /tmp/core.4545 The box was rebooted before I had a change to run gdb, did I miss something? Thank you. Debian normally deletes the contents of /tmp upon boot. -- Tzafrir Cohen icq#16849755jabber:[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] Problem after upgrading from 1.2.21.1 to 1.2.22
On Fri, Jul 20, 2007 at 01:41:42AM -0500, Don Pobanz wrote: You should be running the latest Zaptel LibPRI both of which recently have been updated. We run a similar configuration and have not seen this problem with the upgrade. Even after upgrading Zaptel to the latest (1.2.19) from 1.2.17.1 there is the same problem. Libpri-1.2.5 was already at the lastest. Asterisk-1.2.22 won't work but asterisk-1.2.21 works fine. I still get the following in '/var/log/asterisk/message' pbx.c: Cannot find extension '' in context '(null)' Try enabling debug logging. It may provide more clues. See the sample 'full' in logger.conf . Note that it will provide false alarms. Can you give some context to those messages? from which channel, etc.? Again, nothing has changed in the configuration files. Going back to 1.2.21 corrects the problem. Outgoing calls work fine. Internal calling works fine. (Internal phones are some SIP phones and some Zap phones hanging off of a channel bank) The only lines I am having problems with are our DID trunks incoming from the phone company. They are on a channelized T1 (but not PRI) set up with signalling=em_w in zapata.conf. I have done 'make clean; make install' on zaptel libpri asterisk asterisk-addons Pretty much whenever a new release of asterisk has come out in the 1.2 branch I have updated. (sometimes zaptel gets a little further behind since there is a short outage to upgrade) I have never had any problems before so I believe I am being thorough with the upgrade procedures. Again, has anything changed in how the zapata.conf is parsed? What about extensions.conf when using #includes? I'm not sure what else to do. Does anyone have any other suggestions? I can't think of a better suggestion, so I can only suggest that you try: Install the old version in a chroot or a different system. Copy over all the /etc/asterisk directory. Run in both systems: asterisk -n -rx 'show dialplan' dialplan Are there any changes between the versions? Don Pobanz ___ --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 -- Tzafrir Cohen icq#16849755jabber:[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] G729 copy protection
Jason Parker wrote: I'd wager that you're using the wrong path for the licenses. I believe the correct path is something like /var/opt/asterisk/licenses/ - it's whatever Asterisk has ast_config_AST_VAR_DIR set to, with /licenses/ at the end. The easiest way to tell, is to find the sounds dir (usually at /var/lib/asterisk/sounds/ on Linux), and go up a directory, and then from there create the licenses/ directory. When I register the codec using the register facility, it goes ahead a stores the license file in: /var/lib/asterisk/licenses When I check my asterisk.conf file the location astcarlibdir is as follows: astvarlibdir = /usr/local/asterisk/var/lib I have now tried to symlink the /var/lib/asterisk/licenses to /usr/local/asterisk/var/lib/licenses, and I have also tried to manually create the directory, with the same permissions as the original and copy the license file into the /usr/local/asterisk/var/lib/licenses directory. In each case the asterisk console still comes up with the following error when trying to initialize the codec on startup: Jul 20 08:40:01 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I'm beginning to think that the issue is not the license file, because the above error/warning occurs even when I have not registered the codec. Although, if anyone has more comments/suggestions, please feel free to offer them, I'm willing to try anything twice :) ___ --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] G729 copy protection
Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. ___ --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] Upgrade Procedure
You have to first uninstall your Asterisk1.2 like this-- First you have to stop your asterisk...using-- 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using. 2. After that you have to remove the zaptel driver. For that just run this command and see which are running-- # lsmod | grep zaptel this will show few outputs like this-- zaptel213028 4 zttranscode,ztdummy crc_ccitt 2113 1 zaptel Remove all this as .. modprobe -r zaptel modprobe -r zttranscode . . . . After removing all, again run lsmod | grep zaptel, and you will see nothing. then make clean , all the packages, asterisk, zaptel, and libpri. Then delete the following files, and directory 1. /etc/zapata.conf 2. /etc/asterisk 3./var/lib/asterisk 4./usr/lib/asterisk 5./var/spool/asterisk 6./var/log/asterisk 3. After that install your new Asterisk 1.4 first install zaptel ./configure make menuselect (optional) make make install then Install libpri-- make make install In last install Asterisk package-- ./configure make menuselect (optional) make make install make samples Then start your asterisk , i way you like to it asterisk, or asterisk -g, or sage_asterisk, or asterisk -vvvc, use any of these , of your choice. Regards, Keshav Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh ___ --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 Regards, Kesh Lets change the future...lets change the world. - Need a vacation? Get great deals to amazing places on Yahoo! Travel. ___ --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 use call transfer
There is one thing,
just forget that your phone is a snom phone or whatever...
simply to make a blind call transfer press #8, according to the my
feature.conf, default it is #, or you can assign it any, then after pressing
that you will listen a IVR transfer and dial the desired number followed by
the # sign, then you will connect to the new number, now hangup your phone, and
the other two will be connected.
But make sure, that in your extensions.conf you should have the entry for t,
as I have showed in the entry..
Regards,
Keshav
satish patel [EMAIL PROTECTED] wrote: but what buttons i will use for call
transfer ??? I have SNOM SI 120 phon with transfer button so how to do it ?
Keshav K. [EMAIL PROTECTED] wrote: Hi,
To use call tranfer you have to make entry in extension.conf...
exten = _7.,1,Dial(SIP/${EXTEN},20,Ttr)
then in feature.conf
[featuremap]
blindxfer = #8 ; Blind transfer (default is #)
;disconnect = *0 ; Disconnect (default is *)
;automon = *1 ; One Touch Record a.k.a. Touch Monitor
atxfer = #9; Attended transfer
parkcall = #72; Park call (one step parking)
I'm using this...end its working wonderfully.
--Keshav
satish patel [EMAIL PROTECTED] wrote: Dear all
I have beginer in Voip and i have configured Asterisk server
with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to
transfer call from one user to other means i call to some one and then someone
want to transfer call to another person how it is possible i have also try with
feartue.conf but it is now working i have also read document on voip-info
website but now clear yet can anyone explain me how to asterisk transfer call
from one user to other and what extention.conf look like is there any change in
sip.conf or extention.conf
Rgd
Satish patel
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Kesh
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Re: [asterisk-users] how to use call transfer
Hello,
I'm partner manager at snom. The SI-90 and SI-120 ARE NOT snom phones.
Our Indian joint venture is selling these!
These phones may not be called snom, that's why their name is SI-90 and
SI-120.
The phones are not enginieered and developed in Germany as the normal
snom300 series.
Neither the software nor the hardware was not developed by snom!
I'm not happy with the situation, but I may ask you to directly contact snom
india for support. For me this is a clear bug and should be fixed by them.
I will in parallel contact snom India, so that they take care about the
issue.
In future if you want to get a reliable and working phone, buy a snom from
the actual 300 series. I developed Asterisk solutions myself in the past.
Transfer, Music on Hold, Busy lamp field, etc. all works smooth and hassle
free.
Regards
Tim Koehler
On 7/20/07, Keshav K. [EMAIL PROTECTED] wrote:
There is one thing,
just forget that your phone is a snom phone or whatever...
simply to make a blind call transfer press #8, according to the my
feature.conf, default it is #, or you can assign it any, then after
pressing that you will listen a IVR transfer and dial the desired number
followed by the # sign, then you will connect to the new number, now hangup
your phone, and the other two will be connected.
But make sure, that in your extensions.conf you should have the entry for
t, as I have showed in the entry..
Regards,
Keshav
*satish patel [EMAIL PROTECTED]* wrote:
but what buttons i will use for call transfer ??? I have SNOM SI 120 phon
with transfer button so how to do it ?
*Keshav K. [EMAIL PROTECTED]* wrote:
Hi,
To use call tranfer you have to make entry in extension.conf...
exten = _7.,1,Dial(SIP/${EXTEN},20,Ttr)
then in feature.conf
[featuremap]
blindxfer = #8 ; Blind transfer (default is #)
;disconnect = *0 ; Disconnect (default is *)
;automon = *1 ; One Touch Record a.k.a. Touch Monitor
atxfer = #9; Attended transfer
parkcall = #72; Park call (one step parking)
I'm using this...end its working wonderfully.
--Keshav
*satish patel [EMAIL PROTECTED]* wrote:
Dear all
I have beginer in Voip and i have configured Asterisk
server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how
to transfer call from one user to other means i call to some one and then
someone want to transfer call to another person how it is possible i have
also try with feartue.conf but it is now working i have also read document
on voip-info website but now clear yet can anyone explain me how to asterisk
transfer call from one user to other and what extention.conf look like is
there any change in sip.conf or extention.conf
Rgd
Satish patel
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---
snom technology AG
Tim Koehler
Partner Manager
[EMAIL PROTECTED]
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[asterisk-users] Asterisk Freeze
HI Here is my info: Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my agents are not able to hear any audio only solution is to restart the whole box. Please advise soon. thanks arun ___ --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] No sound from Festival, but *something* is happening
Martin Smith wrote: Hey folks, So I'm trying to get Festival() working on 1.2.17. I'm trying to use app_festival: Here's the show dialplan output from that extension: '3378' = 1. Answer() [pbx_config] 2. Festival(Hello Asterisk caller. How is your day?) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup() [pbx_config] In the Festival server logs, I actually see: client(1) Tue Jul 17 16:38:32 2007 : accepted from localhost client(1) Tue Jul 17 16:38:32 2007 : disconnected But on the channel in question, I hear vm-goodbye and it hangs up. I've turned on the caching option in /etc/asterisk/festival.conf, and then looked in the cache directory, and files *are* appearing there. I'm using the default command: festivalcommand=(tts_textasterisk %s 'file)(quit)\n Even the verbose output shows it working: -- Executing Answer(Zap/97-1, ) in new stack -- Executing Festival(Zap/97-1, Hello Asterisk caller. How is your day?) in new stack == Parsing '/etc/asterisk/festival.conf': Found -- Executing Playback(Zap/97-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') == Spawn extension (default, 3378, 4) exited non-zero on 'Zap/97-1' -- Executing Hangup(Zap/97-1, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'Zap/97-1' -- Hungup 'Zap/97-1' Any ideas as to why I can't hear anything? Thanks! Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ --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 Change: 2. Festival(Hello Asterisk caller. How is your day?) To: 2. Festival(Hello Asterisk caller. How is your day?) You cannot have spaces without quotes. Hope this helps, Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk 1.2.22
On 7/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: Do you have any patches against your Asterisk, Zaptel or Kernel? Actually are you using anything but the factory Kernel? I'm using an older Slackware. The problem came in March or so with 1.2.14 I think. Besides that I just wouldn't advise on using Fedora for any production That's what Mark uses? Course that's not production ;) ___ --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] priorityjumping not working, Dial goes to n+1 not n+101
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087a3000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087b9000 is ringing
-- Nobody picked up in 5000 ms
== Auto fallthrough, channel 'SIP/113-087a3000' status is 'NOANSWER'
With n+1:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,2,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087c8000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087da000 is ringing
-- Nobody picked up in 5000 ms
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/113-087c8000,
SIP/zytek|720|Ttm) in new stack
-- Called zytek
-- Started music on hold, class 'default', on channel
'SIP/113-087c8000'
-- SIP/zytek-087b6000 is ringing
Why?
--
.: Jakub Głazik,
.: email jabber: zytekatnuxi.pl
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Re: [asterisk-users] priorityjumping not working, Dial goes to n+1 not n+101
Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I
assume it was removed from 1.4.
Jakub Głazik wrote
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087a3000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087b9000 is ringing
-- Nobody picked up in 5000 ms
== Auto fallthrough, channel 'SIP/113-087a3000' status is 'NOANSWER'
With n+1:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,2,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087c8000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087da000 is ringing
-- Nobody picked up in 5000 ms
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/113-087c8000,
SIP/zytek|720|Ttm) in new stack
-- Called zytek
-- Started music on hold, class 'default', on channel
'SIP/113-087c8000'
-- SIP/zytek-087b6000 is ringing
Why?
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[asterisk-users] Caller is hanged up after recording voicemail
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
exten = _44950,1,Answer()
exten = _44950,2,Wait,2
exten = _44950,3,Playback(my_Welcome)
exten = _44950,4,Goto(cidchk,s,1) ; we check the availability of the
callerid(num)
exten = 950,1,Background(intro_msg)
exten = 950,2,WaitExten(3)
exten = 950,3,Hangup
exten = 1,1,Goto(csupdesk,s,1)
exten = 2,1,Goto(poll,s,1)
[cidchk]
exten = s,1,Answer()
exten = s,2,GotoIf($[${ISNULL(${CALLERID(num)})} = 1]?nonum,s,1:s,3)
exten = s,3,Goto(pdata,s,1)
[pdata]
exten = s,1,Answer()
exten = s,2,Playback(enter_namenlocation)
exten = s,3,VoiceMail([EMAIL PROTECTED])
exten = s,4,Goto(netchange,950,1)
[poll]
exten = s,1,Answer()
exten = s,2,Playback(short_intro)
exten = s,3,Background(foropinionpoll_press2)
exten = s,4,WaitExten(4)
exten = 2,1,Playback(record_opinion)
exten = 2,2,VoiceMail([EMAIL PROTECTED])
exten = 2,3,Playback(thanku_takingpart)
exten = 2,4,Playback(my_goodbye)
exten = 2,5,Hangup
voicemail.conf
[default]
= 2301,Poll Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
= 1023,Biodata Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
Goksie
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Re: [asterisk-users] G729 copy protection
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) db ___ --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 Channel and VLC
Is it possible to bridge a media stream, lets say created by VLC on to an Asterisk channel? What I would ideally like to do is - When mobile dial into my Asterisk server, follow through some security/prompts then, through the dialplan launch VLC as an external application. VLC would connect to an internet radio service and stream/connect to my channel, when I clear down the VLC app is killed. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip softphone for PDA window mobile 2003 / 5.0 ?
are there any good softphone on PDA window mobile 2003 / 5.0 ? tried sjphone, sound quality is unacceptable. Mario ___ --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] Any plans for proper faxing support
You can also give the our T.38 stack a try. http://www.attractel.com/t38.html Chris Childress AsteriskGuru.com Andrew Joakimsen wrote: I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk hang up the Caller after recording voicemail
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
exten = _44950,1,Answer()
exten = _44950,2,Wait,2
exten = _44950,3,Playback(my_Welcome)
exten = _44950,4,Goto(cidchk,s,1)
exten = 950,1,Background(intro_msg)
exten = 950,2,WaitExten(3)
exten = 950,3,Hangup
exten = 1,1,Goto(csupdesk,s,1)
exten = 2,1,Goto(poll,s,1)
[cidchk]
exten = s,1,Answer()
exten = s,2,GotoIf($[${ISNULL(${CALLERID(num)})} =1]?nonum,s,1:s,3)
exten = s,3,Goto(pdata,s,1)
[pdata]
exten = s,1,Answer()
exten = s,2,Playback(enter_namenlocation)
exten = s,3,VoiceMail([EMAIL PROTECTED])
exten = s,4,Goto(netchange,950,1)
[poll]
exten = s,1,Answer()
exten = s,2,Playback(short_intro)
exten = s,3,Background(foropinionpoll_press2)
exten = s,4,WaitExten(4)
exten = 2,1,Playback(record_opinion)
exten = 2,2,VoiceMail([EMAIL PROTECTED])
exten = 2,3,Playback(thanku_takingpart)
exten = 2,4,Playback(my_goodbye)
exten = 2,5,Hangup
voicemail.conf
[default]
= 2301,Poll Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
= 1023,Biodata Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
Goksie
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[asterisk-users] Which IP Phones will work with non-Asterisk PBX systems too?
Hi everybody, One of my customers wants to buy IP Phones and Asterisk solution, but his requirement is if he'll not be happy with Asterisk, his phones should be able to work with other IP PBX systems as well, so that he doesn't have to buy new phones again. After all IP Phones is the main investment. He'll most probably go with Nortel IP PBX system if he'll be not satisfied with an asterisk system. Experienced folks among you, please advise which phones to offer him, Aastra, Polycom, Snom, or some other. Thanks -- Zeeshan A Zakaria ___ --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 hang up the Caller after recording voicemail
Tell your users to exit voicemail by pressing # instead of hanging up.
Goke Aruna wrote:
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
exten = _44950,1,Answer()
exten = _44950,2,Wait,2
exten = _44950,3,Playback(my_Welcome)
exten = _44950,4,Goto(cidchk,s,1)
exten = 950,1,Background(intro_msg)
exten = 950,2,WaitExten(3)
exten = 950,3,Hangup
exten = 1,1,Goto(csupdesk,s,1)
exten = 2,1,Goto(poll,s,1)
[cidchk]
exten = s,1,Answer()
exten = s,2,GotoIf($[${ISNULL(${CALLERID(num)})} =1]?nonum,s,1:s,3)
exten = s,3,Goto(pdata,s,1)
[pdata]
exten = s,1,Answer()
exten = s,2,Playback(enter_namenlocation)
exten = s,3,VoiceMail([EMAIL PROTECTED])
exten = s,4,Goto(netchange,950,1)
[poll]
exten = s,1,Answer()
exten = s,2,Playback(short_intro)
exten = s,3,Background(foropinionpoll_press2)
exten = s,4,WaitExten(4)
exten = 2,1,Playback(record_opinion)
exten = 2,2,VoiceMail([EMAIL PROTECTED])
exten = 2,3,Playback(thanku_takingpart)
exten = 2,4,Playback(my_goodbye)
exten = 2,5,Hangup
voicemail.conf
[default]
= 2301,Poll Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
= 1023,Biodata Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
Goksie
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Re: [asterisk-users] G729 copy protection
I'd bet the emails are addressed to the list and the original sender, both, so for the original person they appear twice, but everyone on the list gets them a single time. I haven't seen any duplicates. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Friday, July 20, 2007 8:38 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G729 copy protection David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db ___ --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] G729 copy protection
David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db ___ --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] Novice needs help part 2
I am sending an email to the mailer list. Not following any thread? Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, July 20, 2007 2:38 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Novice needs help part 2 Welcome to the Asterisk Users mailing list, On Fri, Jul 20, 2007 at 02:11:08AM -0400, BSumrall wrote: PS, I have zero FX gear. I am 100% SIP Huh? This nmust be related somehow to the PRI Busy problem thread you've answered to, otherwise you wouldn't have replied to it, right? If you want to post a new message, start a new message. Don't just reply to an arbitrary list message and change the subject and contents. Check the list's archives and see that the threading has remained. If you want to follow-up on someone's message, then please reply to it. This will maintain threading. Thus I can easily go one message up and see what signature that PS refers to. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hang up the Caller after recording voicemail
Thank you,
Not that I want the user to hangup...I want the user to continue and
that is why i have the priority 4 on context pdata.
Thanks
Goksie
Eric ManxPower Wieling wrote:
Tell your users to exit voicemail by pressing # instead of hanging up.
Goke Aruna wrote:
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
exten = _44950,1,Answer()
exten = _44950,2,Wait,2
exten = _44950,3,Playback(my_Welcome)
exten = _44950,4,Goto(cidchk,s,1)
exten = 950,1,Background(intro_msg)
exten = 950,2,WaitExten(3)
exten = 950,3,Hangup
exten = 1,1,Goto(csupdesk,s,1)
exten = 2,1,Goto(poll,s,1)
[cidchk]
exten = s,1,Answer()
exten = s,2,GotoIf($[${ISNULL(${CALLERID(num)})} =1]?nonum,s,1:s,3)
exten = s,3,Goto(pdata,s,1)
[pdata]
exten = s,1,Answer()
exten = s,2,Playback(enter_namenlocation)
exten = s,3,VoiceMail([EMAIL PROTECTED])
exten = s,4,Goto(netchange,950,1)
[poll]
exten = s,1,Answer()
exten = s,2,Playback(short_intro)
exten = s,3,Background(foropinionpoll_press2)
exten = s,4,WaitExten(4)
exten = 2,1,Playback(record_opinion)
exten = 2,2,VoiceMail([EMAIL PROTECTED])
exten = 2,3,Playback(thanku_takingpart)
exten = 2,4,Playback(my_goodbye)
exten = 2,5,Hangup
voicemail.conf
[default]
= 2301,Poll Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
= 1023,Biodata Admin,
[EMAIL
PROTECTED],,|attach=yes|saycid=yes|review=yes|sayduration=yes|saydurationm=1
Goksie
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Re: [asterisk-users] G729 copy protection
On Fri, 2007-07-20 at 08:55 -0400, Martin Smith wrote: I'd bet the emails are addressed to the list and the original sender, both, so for the original person they appear twice, but everyone on the list gets them a single time. I haven't seen any duplicates. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Friday, July 20, 2007 8:38 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G729 copy protection David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db Nope, the mails from Bruce are being delivered twice. Yours however only came in once, as do everyone else. So something is strange about the way his emails are encoded I suppose. It isn't really that important to me, but it appeared that Bruce thought he was being slammed for something he wasn't and I wanted to try and let him know he wasn't getting doo. db ___ --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] G729 copy protection
On Thu, Jul 19, 2007 at 02:40:35PM -0800, Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Both copies have the same ID. And both were sent through the gmane newsgroups gatewas. For some reason the order of headers lsightly differs in the Newsgroup headers appears only in one of the two. -- Tzafrir Cohen icq#16849755jabber:[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] Novice needs help part 2
On Fri, Jul 20, 2007 at 09:01:27AM -0400, BSumrall wrote: I am sending an email to the mailer list. Not following any thread? Brad This email was a reply to my message, and hence appeared properly threaded to it. Look for BSumrall in http://lists.digium.com/pipermail/asterisk-users/2007-July/thread.html (how nice it is to be able to edit threads with mutt) -- Tzafrir Cohen icq#16849755jabber:[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] G729 copy protection
Bruce sorry for the top post, but your last two messages have not come in twice Go figure... db On Fri, 2007-07-20 at 13:37 +0100, Bruce McAlister wrote: David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db ___ --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] Which IP Phones will work with non-Asterisk PBX systems too?
Hi, how compatible a phone is to a specific phone system also depends from the required feature set. Basic call should work between most phones and systems. If you're interested on snom interoperability I would like to point your attention to the following page: http://webcms.snom.com/wiki/index.php/Interoperability Regards Tim On 7/20/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi everybody, One of my customers wants to buy IP Phones and Asterisk solution, but his requirement is if he'll not be happy with Asterisk, his phones should be able to work with other IP PBX systems as well, so that he doesn't have to buy new phones again. After all IP Phones is the main investment. He'll most probably go with Nortel IP PBX system if he'll be not satisfied with an asterisk system. Experienced folks among you, please advise which phones to offer him, Aastra, Polycom, Snom, or some other. Thanks -- Zeeshan A Zakaria ___ --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 -- --- snom technology AG Tim Koehler Partner Manager [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] Unicall/R2 for Asterisk 1.4 Available for TESTING
Only to continue on this thread (becouse this is start in other meail). The 1.4.X. unicall patch is working well, only with one problem: There is a problem hen reciving calls with no Caller ID. Thanks. On 6/9/07, Moises Silva [EMAIL PROTECTED] wrote: Alvaro... Hum..., I never have tried RxFax... let me know if you need any extra help with that. Sounds interesting On 6/8/07, Alvaro Parres [EMAIL PROTECTED] wrote: Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke every time... With and error in the linking to the spandsp library. If i have time this weekend i will review to fix the app, Thanks. On 6/4/07, Tobias Wolf [EMAIL PROTECTED] wrote: Humberto Figuera schrieb: HI Tobias, look in www.soft-switch.org/unicall/unicall/index.html ;p Thank you. Not very complete but it has given me an idea what to think of unicall. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [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] Asterisk 1.4, Unicall and Nextel...
Search at mfcr2.c this: case MFCR2_PROT_MEXICO: And add the next line after that line: mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12; This will help you on calls that have the restricted flag on the ANI only. (Nextel). But not on no caller id calls. I don't know if steve can help us whit the case where no caller id is send. On 7/19/07, Carlos Chavez [EMAIL PROTECTED] wrote: *On Thu, 19 Jul 2007 12:14:53 -0500, Alvaro Parres wrote* Yes Moises, i was looking for it. The main problem is only on the files for version 1.4... it give that error when no CallerID is recive or a private caller id is recive. The change i made is to add to Mexico variant on mfcr2.c this line mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12; This works for nextel or phones that send private caller id.. But doesn't work when no CallerID is recive. I have al ready check diff files from 1.2 files and 1.4 files and i didn't find any big difference between both version. Ok, I did the change you specified and now we can receive calls from Nextel phones but get no callerid on any call. How do I apply the patch to libmfcr2? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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 -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More list issues [Re: G729 copy protection]
On Fri, 20 Jul 2007, Martin Smith wrote: I'd bet the emails are addressed to the list and the original sender, both, so for the original person they appear twice, but everyone on the list gets them a single time. I haven't seen any duplicates. I've seen list duplicated and sometimes triplicates here, and as someone who runs many mailing lists myself, I often see duplicates on my lists. It's very rarely the original sender at fault, and vary rarely the list processor. IME it's mostly caused by broken MS Exchange servers somewhere which a list member is using, which feed messages back into the list with the same message-id. They usually clear themselves up, but sometimes I have to dig out the offending site and let them know. Gordon ___ --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 novice needs help.
On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: My dial plan of issues….. exten = s,1,Answer(60) exten = s,2,Background(otherwise-press) exten = s,1,Playback(digits/1) exten = s,2,Goto(default,s,1) exten = s,1,Playback(digits/2) exten = s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two here... Also, you have a *very* long argument to the Answer() application. Usually a second or two is plenty. Try something like this: exten = s,1,Answer(1) ; answer the call, then wait 1 second ; before going on to the next priority exten = s,2,Background(vm-enter-num-to-call) ; play prompt in ; background, waiting for caller to ; enter DTMF digits exten = s,3,WaitExten(); continue to wait for digits after the ; prompt has finished exten = 1,1,SayDigits(1) ; say one exten = 1,2,Goto(s,1) ; go back to the menu exten = 2,1,SayDigits(2) ; say two exten = 2,2,Goto(s,1) ; go back to the menu Hopefully that will get you started in the right direction. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow list
Hi Philipp - Since the list was switched over to API-Digital almost every message I get is older than a week. Coincidence? Is anyone else having trouble? Well, this is now the third active thread on this subject, but I guess you won't see this message for a while. Has anyone dissected the headers of a delayed message yet? We should be able to tell for sure where the holdup is. All of the messages are coming through on time for me, so it won't do much good for me to look. - Noah ___ --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 IVR Performance
I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe something like FastAGI) Right now I'm writing in a scripting language, would there be a performance gain from writing in a compiled language? I don't see any serious memory utilization and normally processor utilization is below 50% with spikes to 70% under load with four or five ExternalIVRs running. I will gladly provide any additional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] Any plans for proper faxing support
At 07:22 7/20/2007, Chris Childress wrote: You can also give the our T.38 stack a try. http://www.attractel.com/t38.html Software? Hardware? Integration? Prices? Can't make a decision without enough info. Chris Childress AsteriskGuru.com Andrew Joakimsen wrote: I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. ___ --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] G729 copy protection
Not until the topic looks me in the face. Sorry I wasn't up earlier in the morning to explain my question, but the rest of the list did for me. I was seeing two messages _every_ time you posted one. Like I said, I was just curious if that could have been the reason; it would have been an easy fix ;) No offense intended. Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. ___ --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] Sip Providers
Hi Anthony, I used their services for 3 months (signed up on the pay-as-you-go plan - Unlimited channels) and all my minutes were rounded up to the next cent in my CDR ... Regards, Marcelo -- Original message -- From: Al Bochter [EMAIL PROTECTED] Anthony, So you know all 4 that work at teliax.com I only know what others have told me about teliax.com Most of what I know was told to me from someone that worked there. Best regards, Al Bochter http://www.BochterServices.com --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Anthony Francis wrote: Darrick Hartman (lists) wrote: [EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. I also vouch for Teliax as I send overflow LD through their trunks. I know the people there and they are great guys. ___ --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 Inbound (clean). Database: 000757-4, 07/18/2007 - 7/19/2007 10:35:00 AM ---BeginMessage--- ___ --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---End Message--- ___ --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] * core file not recognized
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: /tmp/core.4545 is not a core dump: File format not recognized So what is that file? file core.4545 core.4545: ELF 32-bit LSB core file Intel 80386, version 1 (SYSV), SVR4-style, SVR4-style, from 'asterisk' The box was rebooted before I had a change to run gdb, did I miss something? Thank you. Debian normally deletes the contents of /tmp upon boot. I copied the file off and copied it back in case gdb wanted it in temp. Thanks, JJ Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --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] priorityjumping not working, Dial goes to n+1 not n+101
On 7/20/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I assume it was removed from 1.4. According to UPGRADE.txt, the default in the absence of priorityjumping= changed from yes in 1.2 to no in 1.4: * In previous Asterisk releases, many applications would jump to priority n+101 to indicate some kind of status or error condition. This functionality was marked deprecated in Asterisk 1.2. An option to disable it was provided with the default value set to 'on'. The default value for the global priority jumping option is now 'off'. But there is no indication that they removed the option to have priorityjumping turned on globally. -- 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] ulaw to g726 conversion
I am able to use sox to convert audio files from ulaw to wav (MS ADPCM), is there a way, using sox or another command line tool, to convert them to g726 ? ( g726-32 since it is supported by * ) tia, -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] In Vancouver is it a local to call from 778 to 604, and vice versa?
Yes it is a local call. 778 was added to the lower mainland to ease the pressure on 604 numbers due to the explosion in mobile/fax lines being added. If you need a 604DID (for vanity purposes or whatever) you could try the provider I use. I'm not sure if I can post their information on the list (advertising and all) so feel free to email me personally and I'll put you in touch with them. Mike Wood BC Northern Lights 1-866-933-3269 ext 113 1-604-543-1768 (fax) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Wednesday, July 18, 2007 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] In Vancouver is it a local to call from 778 to 604,and vice versa? I've got a 778 DID for vancouver, but don't know if it will be a local call if dialed 604 and vice versa. What are the different area codes in Vancouver and why its easier to get 778 DID than 604? -- Zeeshan A Zakaria No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.9/907 - Release Date: 7/18/2007 3:30 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.9/907 - Release Date: 7/18/2007 3:30 PM ___ --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] Problem
i am have x100P clone, and install asterisk 1.4 and out call normaly and hangup in xlite to zap but call to asterisk for zap channel nop pass to xlite and the caller hangup the asterisk not detect. what is the problem ??? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem
On Fri, 2007-07-20 at 15:33 -0500, Walter Willis wrote: i am have x100P clone, and install asterisk 1.4 and out call normaly and hangup in xlite to zap but call to asterisk for zap channel nop pass to xlite and the caller hangup the asterisk not detect. The X100P (and it's numerous clones) don't support far-end disconnect supervision. This means they can't tell when the far end has hung up the call. I suggest you might want to try a more modern telephony card, such as the Digium TDM400P card. The X100P cards are just more headache than they're worth (and have been since I started using Asterisk over 5 years ago). -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing Digium/Asterisk World's Conference Program
Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean From: Carl Ford [mailto:[EMAIL PROTECTED] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program http://www.magnet101.com/ls.cfm?r=49686240sid=2533069m=336237u=Pulve rmedis=https://secure.pulver.com/digiumAsteriskWorld/2007/boston/web/at tendRegister.htm Hi Dean, I am pleased to announce that the conference program http://www.magnet101.com/ls.cfm?r=49686240sid=2533070m=336237u=Pulve rmedis=http://www.digiumasteriskworld.com/2007/boston/web/confSchedule. htm for Digium /Asterisk World http://www.magnet101.com/ls.cfm?r=49686240sid=2533071m=336237u=Pulve rmedis=http://www.digiumasteriskworld.com/2007/boston/web/ is now posted to the website! Click here http://www.magnet101.com/ls.cfm?r=49686240sid=2533072m=336237u=Pulve rmedis=https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07ty pe=gpricode=emcf718 to register today! Digium Asterisk World is the premier open communication event for the channel that encompasses the world of open source platforms and applications in the realm of IP communications. Whether you are a service provider, VAR, systems integrator or someone who is rolling out IP communications internally, Digium Asterisk World is the place to be. Digium Asterisk World will be held October 30 - 31, 2007 at the Boston Conference and Convention Center in Boston, MA. Please visit http://www.digiumasteriskworld.com http://www.magnet101.com/ls.cfm?r=49686240sid=2533073m=336237u=Pulve rmedis=http://www.digiumasteriskworld.com/ for complete details. As a member of the VON Family, we have developed a special discount of 50% off of the conference for you, if you register by July 29, 2007. To take advantage of this limited time offer, please register here http://www.magnet101.com/ls.cfm?r=49686240sid=2533074m=336237u=Pulve rmedis=https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07ty pe=gpricode=emcf718 ! Also, please feel free to pass this on to your colleagues who may be interested as well. Sincerely, Carl Ford VP of Content Community Pulvermedia P.S. Remember as a member of the VON Family, we have developed a special discount of 50% off of the conference for you, if you register by July 29, 2007. To take advantage of this limited time offer, please register here http://www.magnet101.com/ls.cfm?r=49686240sid=2533075m=336237u=Pulve rmedis=https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07ty pe=gpricode=emcf718 ! http://www.pulver.com/emails/daw/2007/boston/images/rightTopCurve.gif http://www.pulver.com/emails/von/2007/boston/images/headRight_special.g if Save 50% off of the conference! Register by July 29, 2007 http://www.magnet101.com/ls.cfm?r=49686240sid=2533076m=336237u=Pulve rmedis=https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07ty pe=gpricode=emcf718 http://www.pulver.com/emails/daw/2007/boston/images/headRight_bottom.gi f http://www.pulver.com/emails/daw/2007/boston/images/headRight_premier.g if http://www.pulver.com/emails/daw/2007/boston/images/sponsors_premier.gi f http://www.pulver.com/emails/daw/2007/boston/images/headRight_bottom.gi f http://www.pulver.com/emails/daw/2007/boston/images/headRight_platinum. gif http://www.pulver.com/emails/daw/2007/boston/images/sponsors_platinum.g if http://www.pulver.com/emails/daw/2007/boston/images/headRight_bottom.gi f http://www.pulver.com/emails/daw/2007/boston/images/rightBottom.gif http://www.magnet101.com/ls.cfm?r=49686240sid=2533077m=336237u=Pulve rmedis=https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07ty pe=gpricode=emcf718 To unsubscribe from Digium Asterisk World emails please click here http://www.magnet101.com/ls.cfm?r=49686240sid=2533078m=336237u=Pulve rmedis=http://www.digiumasteriskworld.com/2007/boston/web/unsubscribe.h tml To unsubscribe from future Pulvermedia emails please click here http://www.magnet101.com/ls.cfm?r=49686240sid=2533079m=336237u=Pulve rmedis=http://www.pulver.com/unsubscribe/ http://www.magnet101.com/spacer.cfm?tracking_id=1210030106_Pulvermedi ___ --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 novice needs help.
Jared Smith wrote: On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: My dial plan of issues….. exten = s,1,Answer(60) exten = s,2,Background(otherwise-press) exten = s,1,Playback(digits/1) exten = s,2,Goto(default,s,1) exten = s,1,Playback(digits/2) exten = s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two here... Also, you have a *very* long argument to the Answer() application. Usually a second or two is plenty. Try something like this: According to Asterisk 1.2.17 the delay is in milliseconds or, in this case, 0.06 seconds. ... asterisk*CLI show application Answer asterisk*CLI -= Info about application 'Answer' =- [Synopsis] Answer a channel if ringing [Description] Answer([delay]): If the call has not been answered, this application will answer it. Otherwise, it has no effect on the call. If a delay is specified, Asterisk will wait this number of milliseconds before returning to the dialplan after answering the call. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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] Announcing Digium/Asterisk World's Conference Program
Dean Collins wrote: Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. No it is not, as is clearly evidenced by the fact that the Astricon website is still up, with the same content it had before this announcement, and still accepting registrations :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] Aastra phones loosing service...
I have a customer that has recently upgraded their network and now their Aastra 9133i phones are loosing their connection to the Asterisk server. They were using an external Asterisk server and now we have installed a new internal server with Asterisk 1.4.8 on a SIP/IAX implementation with no Zap cards. Only hard phones seem to be having problems keeping a connection to the server, soft phones do not seem to be affected. I was wondering if anyone here has had an experience where hard phones have problems connecting to the server. The client has good Cisco intelligent switches on their network. Is it possible that they maybe sending some QoS or VLAN information that confuses the phones? -- 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] Announcing Digium/Asterisk World's Conference Program
On Fri, 2007-07-20 at 17:01 -0400, Dean Collins wrote: Is this replacing Astricon this year? Nope... this isn't meant to replace AstriCon in the slightest. As I understand it, the two conferences have different focuses... Obviously AstriCon is the Asterisk users' conference and is meant to cover *everything* related to Asterisk (and is more likely to be attended by the Asterisk faithful). On the other hand, Digium/Asterisk World is targeted at the unwashed masses -- businesses and resellers that want to look at an open source alternative but are not converted to our brand of telecom religion. In short, we hope to see you at AstriCon, whether or not you choose to check out Digium/Asterisk World! -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * core file not recognized
On Fri, Jul 20, 2007 at 10:09:33AM -0700, Jay Wilton wrote: --- Tzafrir Cohen [EMAIL PROTECTED] wrote: /tmp/core.4545 is not a core dump: File format not recognized So what is that file? file core.4545 core.4545: ELF 32-bit LSB core file Intel 80386, version 1 (SYSV), SVR4-style, SVR4-style, from 'asterisk' What was the command-line you used with gdb? The box was rebooted before I had a change to run gdb, did I miss something? Thank you. Debian normally deletes the contents of /tmp upon boot. I copied the file off and copied it back in case gdb wanted it in temp. gdb doesn't really care. -- Tzafrir Cohen icq#16849755jabber:[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] Problem
On Fri, Jul 20, 2007 at 04:56:45PM -0400, Jared Smith wrote: On Fri, 2007-07-20 at 15:33 -0500, Walter Willis wrote: i am have x100P clone, and install asterisk 1.4 and out call normaly and hangup in xlite to zap but call to asterisk for zap channel nop pass to xlite and the caller hangup the asterisk not detect. The X100P (and it's numerous clones) don't support far-end disconnect supervision. This means they can't tell when the far end has hung up the call. What far-end disconnect supervision doesn't it support? It supports power denian - KS (or else the wcfxo is simply lying), which is the common method in the US. It doesn't support other metheds such as polarity reversal. Anyway, a common workaround is to use (yuck) busydetect - detecting a busy tone on the line. -- Tzafrir Cohen icq#16849755jabber:[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] POE injector
I'm looking for 24 or 48 port IEEE802.3af POE injector. Any recommendation? ___ --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] ulaw to g726 conversion
On 7/20/07, Thomas Kenyon wrote: convert file.g729 file.g726-32 in the asterisk CLI works here. as does file.g726-16 (but not 24 or 40). The weird thing is, it doesn't seem to transcode from ulaw/alaw but works fine from g729/gsm. thank you ! now I have another command to experiment with. The downside to converting from g729 / gsm to g726 is that they are lossier than g726, which is richer in information. Ideally convert from ulaw to everything else. thnx again. -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] Has anybody used fanless computers of logic supply with asterisk?
Hi, I have to install an Asterisk PBX for a customer and he wants something like logic supply's fanless computers. Can anybody advise about how good will they work, are they compatible with the Asterisk system? I'll also be installing a sangoma 4 port FXO card in it. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem
look my zapata.conf [channels] context=default switchtype=national signalling=fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.0 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=3 answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=1 callprogress=no musiconhold=default channel = 1,2 add to line: busydetect=yes busycount=3 but the situation is iqual have much that it is a clon x100p ??? # ztcfg -v Zaptel Version: 1.4.4 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. it is in 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
[asterisk-users] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D card.
To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup.
So I wanted to know the steps any issue which I may come accross if any.
I have googled have some docs handy wrt Trixbox-2.2. Just wanted to get some
notes from user with custom install setup when used with
Asterisk+freepbx+Sangoma.
Also how do I enable DTMF hardware detection.
--
Deepak
Linux your Life, Don't Window it [[]]
{ All for the best }
-
Yahoo! Answers - Get better answers from someone who knows. Tryit now.___
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Re: [asterisk-users] Aastra phones loosing service...
On 7/20/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that has recently upgraded their network and now their Aastra 9133i phones are loosing their connection to the Asterisk server. They were using an external Asterisk server and now we have installed a new internal server with Asterisk 1.4.8 on a SIP/IAX implementation with no Zap cards. Only hard phones seem to be having problems keeping a connection to the server, soft phones do not seem to be affected. I was wondering if anyone here has had an experience where hard phones have problems connecting to the server. The client has good Cisco intelligent switches on their network. Is it possible that they maybe sending some QoS or VLAN information that confuses the phones? I have seen the issue with the Aastra phones they periodically loose connection and then a little while after gain connection again. At the end of the day they have poor support, as in Maybe we'll fix it in a subsequent release ___ --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] pattern base call routing
exten = _98XX,1,Dial(ZAP/(your preferred E1) exten = _,1,Dial(ZAP/(second E1) On 7/20/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have 2 E1 card on my asterisk and i want to route call with fix pattern like if anyone dial mobile number like 9818875535 so it will use PRI 1 and rest of the world goes through PRI 2 means whn number prefix 98XX then call goes through specified E1 is it possible ??? satish patel -- Get the free Yahoo! toolbarhttp://us.rd.yahoo.com/evt=48226/*http://new.toolbar.yahoo.com/toolbar/features/norton/index.phpand rest assured with the added security of spyware protection. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 61
Please, unsuscriber, this group. regars Nestor Castillo - Mensaje original De: [EMAIL PROTECTED] [EMAIL PROTECTED] Para: asterisk-users@lists.digium.com Enviado: viernes, 20 de julio, 2007 11:00:04 Asunto: asterisk-users Digest, Vol 36, Issue 61 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: asterisk novice needs help. (Jared Smith) 2. Asterisk IVR Performance (David Ruggles) 3. Re: Any plans for proper faxing support (Doug) 4. Re: G729 copy protection (Mojo with Horan Company, LLC) -- Message: 1 Date: Fri, 20 Jul 2007 10:30:17 -0400 From: Jared Smith [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk novice needs help. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: My dial plan of issues?.. exten = s,1,Answer(60) exten = s,2,Background(otherwise-press) exten = s,1,Playback(digits/1) exten = s,2,Goto(default,s,1) exten = s,1,Playback(digits/2) exten = s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two here... Also, you have a *very* long argument to the Answer() application. Usually a second or two is plenty. Try something like this: exten = s,1,Answer(1) ; answer the call, then wait 1 second ; before going on to the next priority exten = s,2,Background(vm-enter-num-to-call) ; play prompt in ; background, waiting for caller to ; enter DTMF digits exten = s,3,WaitExten(); continue to wait for digits after the ; prompt has finished exten = 1,1,SayDigits(1); say one exten = 1,2,Goto(s,1); go back to the menu exten = 2,1,SayDigits(2); say two exten = 2,2,Goto(s,1); go back to the menu Hopefully that will get you started in the right direction. -- Jared Smith Community Relations Manager Digium, Inc. -- Message: 2 Date: Fri, 20 Jul 2007 12:13:06 -0400 From: David Ruggles [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk IVR Performance To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;charset=us-ascii I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe something like FastAGI) Right now I'm writing in a scripting language, would there be a performance gain from writing in a compiled language? I don't see any serious memory utilization and normally processor utilization is below 50% with spikes to 70% under load with four or five ExternalIVRs running. I will gladly provide any additional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -- Message: 3 Date: Fri, 20 Jul 2007 11:46:58 -0500 From: Doug [EMAIL PROTECTED] Subject: Re: [asterisk-users] Any plans for proper faxing support To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed At 07:22 7/20/2007, Chris Childress wrote: You can also give the our T.38 stack a try. http://www.attractel.com/t38.html Software? Hardware? Integration? Prices? Can't make a decision without enough info. Chris Childress AsteriskGuru.com Andrew Joakimsen wrote: I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. ___ --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] Has anybody used fanless computers of logic supply with asterisk?
Hi Zeeshan - I have to install an Asterisk PBX for a customer and he wants something like logic supply's fanless computers. Can anybody advise about how good will they work, are they compatible with the Asterisk system? I'll also be installing a sangoma 4 port FXO card in it. Have you thought about using Digium's new Asterisk Appliance? It has up to eight analog ports and comes with hardware echo cancellation and ABE. - Noah ___ --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] ulaw to g726 conversion
On 7/20/07, I wrote: On 7/20/07, Thomas Kenyon wrote: convert file.g729 file.g726-32 in the asterisk CLI works here. as does file.g726-16 (but not 24 or 40). The weird thing is, it doesn't seem to transcode from ulaw/alaw but works fine from g729/gsm. thank you ! now I have another command to experiment with. The downside to converting from g729 / gsm to g726 is that they are lossier than g726, which is richer in information. Ideally convert from ulaw to everything else. just to clarify, I meant convert from the highest definition codec you have ( alaw, ulaw, slin ) to better compression codecs ( g726, g729, gsm ). -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] Redundancy / Failover
Hi Norman - To add to what Edgar said, yes, use linux-ha. It works nicely in combination with DRBD. DRBD uses a dedicated network interface on each box with a crossover cable between the two. It does a block level copy of the entire filesystem, so you have two machines that are identical. The you use the linux-ha heartbeat to monitor the OS and asterisk. If anything goes wrong, it can fail over to the second machine. This is pretty easy to set up with Analog lines. With PRI's you'd need the fonebridge or the FSV-4PFS from http://www.failsafevoip.com Thanks, I wasn't aware of the FSV-4PFS box. Can one switch it remotely (e.g. over the network?) From what I understand, it has its own heartbeat-type monitoring of asterisk. If asterisk fails, it will automatically fail the PRI over to your backup machine. Can you manually force the failover? I think so, but I'm not positive. You can ask the failsafevoip people directly. I've exchanged emails with them before and they are knowledgeable and responsive. It would be nice to have a way to gracefully switch boxes, e.g. all new calls to the backup box, wait until all calls on the primary normally end, and then take server down for an upgrade. If you're using heartbeat, and it's directly monitoring the asterisk process, you should be able to issue a stop gracefully command. That will bring asterisk down when all the calls are complete. Then, heartbeat should fail over to the other machine. Of course, if someone is on a long call and you've already issued a stop gracefully command, your asterisk cluster won't accept any new calls until that long call is finished. - Noah ___ --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 Freeze
Hi Arun - Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my agents are not able to hear any audio only solution is to restart the whole box. Please advise soon. You really need to update to a later version of asterisk (and zaptel). There have probably been somewhere close to a thousand bug fixes since 1.2.10. If you still have this issue with the latest version, please collect as much information as possible (exact cli messages, turn on logging, your config files, etc) and post that information to this list. - Noah ___ --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 use call transfer
If you're using the Snom transfer button, you don't need to do anything in features.conf. In extensions.conf, just make sure that the dial() command used to call the snom phone uses the 't' flag. THIS IS INCORRECT! The options t and T are for DTMF based transfers. You do not need any options to Dial() to do phone based transfers using the transfer button on your IP phone (or FLASH on your IP ATA). Yes, Eric's answer is correct, mine is incorrect. My bad. (I forgot as I usually use the DTMF-based transfer). - Noah ___ --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] POE injector
I'm looking for 24 or 48 port IEEE802.3af POE injector. Any recommendation? Yes. For the price of one of those multi-port injectors, you can come close to the price of a new Netgear or 3Com PoE switch. The injectors typically add power to the unused pairs (mode B PoE). This means you can't use them on anything better than fastethernet. When switches do PoE natively, they put the power on the data carrying pairs (mode A PoE), so they can do gigabit ethernet. I think PowerDsine makes a PoE injector that uses mode A, and so it can do gigabit ethernet. - Noah ___ --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] Upgrade Procedure
You have to first uninstall your Asterisk1.2 like this-- First you have to stop your asterisk...using-- 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using. In my experience, you don't need to do this step. In fact, you can keep the old asterisk running, compile and install asterisk 1.4 on top of it. Then issue a restart when convenient command from the asterisk 1.2 prompt, and Asterisk 1.4 will come up after the restart. - Noah ___ --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] POE injector
IEEE802.3af uses same 4 wire as data. thats what Polycom uses. the way i'm seeing it we are better off with poe switch(looking at the price). On 7/20/07, Noah Miller [EMAIL PROTECTED] wrote: I'm looking for 24 or 48 port IEEE802.3af POE injector. Any recommendation? Yes. For the price of one of those multi-port injectors, you can come close to the price of a new Netgear or 3Com PoE switch. The injectors typically add power to the unused pairs (mode B PoE). This means you can't use them on anything better than fastethernet. When switches do PoE natively, they put the power on the data carrying pairs (mode A PoE), so they can do gigabit ethernet. I think PowerDsine makes a PoE injector that uses mode A, and so it can do gigabit ethernet. - Noah ___ --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
