Re: [asterisk-users] Enterprise or Fedora?
I have been running FC5 for 3 years with no issues, I also started using for FC7 the last 2 months. I have not had any issues. On 2/3/08, Goke Aruna [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote: Ubuntu server for me please simply, is better... install. then activate universe and multiverse repositories sudo apt-get update sudo apt-get upgrade sudo apt-get build-dep asterisk and then... tar xvfz ./configure make make install ... is very very easy and clean, and IMHO i guess is better SO Ubuntu than any other RHEL based distro... Mind you, asterisk is in the ubuntu universe (rather than main) archive. As such, it also has some other dependencies in universe. Specifically (looking at the current Hardy) package: libc-client2007-dev libiksemel-dev libopenh323-dev libpri-dev libradiusclient-ng-dev libtonezone-dev libvpb-dev zaptel-source As a rule, the universe packages there get much less attention, and don't necessarily get regular security updates. i think, its matter of knowing what you want, I have been running FC5 with sangoma a104dx card for over a year, i have not experienced any serious stress. goksie ___ -- 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] Multiple SIP phones behind a Linksys firewall
My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. I have seen this before on badly behaved home routers that have a hidden SIP Proxy, notably Zyxel wireless units. I've not seen it happening on either Linksys or Netgear units though. Do you actually need STUN? In my experience it can cause more problems than it solves, especially if the public IP changes and the STUN server isn't due to be queried for another X seconds. If possible, and assuming it won't create unreasonable load on your * server, try dropping the registration interval down to something small like 300 (5 minutes), and disable STUN entirely (obviously making sure nat=yes is defined in sip.conf for those devices). Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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] X-Lite Softphone keeps de-registering?
Hello, On Feb/01/2008, Doug wrote: The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? You can also use OpenVPN (using port 80, I don't know if it's possible to use TCP, I think so). So all SIP and RTP traffic would use it. -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- 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] Telco MWI Detection on TDM400 Interface?
I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming channel = 4 However, the zapnotify.sh script in the /usr/local/sbin directory is never getting called. Do I need a new version of the zaptel drivers? I'm currently using version 1.4.6 Is there anyway to get debugging into the log files? Am I missing something? Jim ___ -- 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] Multiple SIP phones behind a Linksys firewall
Do you have a range of registration ports configured and forwarded through the firewall on the server end? Ie. 5060-5065 for example. On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc. and configure the phones to use that port for registration. You may need to forward ports for the actual voice as well. 2 ports per phone so 1-10001 for phone1 and 10002-10003 for phone2. It's either that or mess around with STUN or Proxy servers or whatever. SIP+NAT=headache -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, February 02, 2008 8:23 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall The server is at a remote datacenter - no nat, no firewall, pure public IP. The phones are at home offices (i.e. DSL or Cable with Linksys-type firewall/routers). My initial testing was with a single SIP phone at the home office - and everything worked fine. But when I have two SIP phones at the home office, things start behaving badly. I understand the issue of phone-to-phone, where both phones are behind a nat at the home office - but that is not the issue I am having. My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. ___ -- 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] switch QOS requirements
Dear List, We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Test Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switch QOS requirements
John Williams [EMAIL PROTECTED] writes: We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks If you have one switch for the whole network, you're generally fine without QoS. Switches these days can handle full bandwidth on all ports at the same time. Anyway, Asterisk is no different from other PBX's when it comes to QoS. Should it turn out that you actually need it on the LAN, just be sure you set the tos parameters in sip.conf to something that is prioritized by the switch. /Benny ___ -- 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] QueueMember event/LastCall Variable - Format?
Lee Jenkins wrote: Jared Smith wrote: On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I recall. Thanks. Apparently Asterisk reports it in QueueMember* AMI events as GMT based as well, requiring that we apply our regional offsets (GMT -5 for instance). -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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] Enterprise or Fedora?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, Just to throw in my 2 cents worth, I have been evaluating most of the 'big' distro's for my Asterisk platform and since I've used almost all of them I kept the pre-requisites to these standard options. 1. Package management - easy to administer, easy to upgrade and include development packages 2. System stability - Gotta be rock solid 3. Proprietary Code - It must be all open source 4. Standards Compliant With a lot of testing, compiling and downloading I came to the conclusion that the winner was Debian 'Etch' Reason: 1. Apt - The best package management system ever. 2. Rock solid platform, my other Debian servers have been running for years (and so have my CentOS servers so I'm not being biased) 3. The software is all open source. 4. I have a requirement for H323 in the vast majority of my asterisk boxes and it compiles brilliantly with Etch 5. 1 CD install - The Net-Install CD makes things neat 6. Slow release cycle and easy upgrade to latest version once released. CentOS came a very close second however it was the apt difference that clinched it for Debian. I would have went Ubuntu instead of Debian however the packages are not quite as concise and I'm still not sure on the whole 'sudo' method of security, its great for a desktop but the jury is still out on a server. Regards, Alan love U.all wrote: | i wanna build a production Asterisk box ,will RedHat Linux Enterprise | Server be more stable than Fedora core Linux or it makes no significant | difference | | Express yourself instantly with MSN Messenger! MSN Messenger | http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ | | | | | ___ | -- 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 - -- Alan Hanley FSF Member 4949 Windows is a half-baked, dying OS that in essence is a 32 bit extension and graphical shell, for a 16 bit patch to an 8 bit operating system, originally coded for a 4 bit microprocessor, written by a 2 bit company, that can't stand 1 bit of competition. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHpkl3MRbQISW5zV8RAoBLAJ9Ga/dGRyZkQeBwrBNTVkIA5+jZtwCeLBO8 xNEKOPuH8t+lzRG7OUb7ChI= =lAyF -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Console/dsp, makes me sound like a Dalek
I need to set up the sound card of a server to use in an overhead paging system, as normal I am testing this on my home machine first (which has slightly different Hardware). I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G (ICH7 Family) sound card. I am running Asterisk 1.4.17 and have a fully loaded TDM400P as a timing source. When calling console/dsp (using various methods) the output from the sound card changes the voice to sound like a Dalek, which looks similar to bug id# 0003005 from 2004, (to which the answer was to buy a better card). The server that I will need to get this running on has an 82801EB/ER (ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put another card in). Does anybody know if this can work with the above hardware? Does anybody know where I should start looking to fix this? TIA for any help with this matter. ___ -- 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] Console/dsp, makes me sound like a Dalek
Thomas Kenyon wrote: When calling console/dsp (using various methods) the output from the sound card changes the voice to sound like a Dalek, which looks similar When this happens to me, I switch. If is sounds badly with Alsa, try OSS. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console/dsp, makes me sound like a Dalek
I thought sounding like a dalek was a good thing. PaulH On Sun, 2008-02-03 at 23:56 +, Thomas Kenyon wrote: I need to set up the sound card of a server to use in an overhead paging system, as normal I am testing this on my home machine first (which has slightly different Hardware). I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G (ICH7 Family) sound card. I am running Asterisk 1.4.17 and have a fully loaded TDM400P as a timing source. When calling console/dsp (using various methods) the output from the sound card changes the voice to sound like a Dalek, which looks similar to bug id# 0003005 from 2004, (to which the answer was to buy a better card). The server that I will need to get this running on has an 82801EB/ER (ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put another card in). Does anybody know if this can work with the above hardware? Does anybody know where I should start looking to fix this? TIA for any help with this matter. ___ -- 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] AGI: Not getting answers from get_data in a call-file call
I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); $AGI-answer(); my $i; $i = $AGI-channel_status(); $AGI-say_digits($i); $i = $AGI-get_data(one-moment-please, 1, 3); $AGI-say_digits($i); As you can see, nothing serious. When running this script in a normal telephone call, it works. When running this script in the call created with the call-file, I do hear the output of the first say_digits and the one-moment-please, but the pressing of the DTMF keys is not recognized by the system, the get_data() times out and the function returns nothing. This script once worked in 1.2.x (It was part of a bigger project) but now that I want to move it to 1.4 it gives this strange behaviour. Is there anybody with a hint on how to resolve this? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ -- 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] switch QOS requirements
On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote: John Williams [EMAIL PROTECTED] writes: We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks If you have one switch for the whole network, you're generally fine without QoS. Switches these days can handle full bandwidth on all ports at the same time. Anyway, Asterisk is no different from other PBX's when it comes to QoS. Should it turn out that you actually need it on the LAN, just be sure you set the tos parameters in sip.conf to something that is prioritized by the switch. It tends to be more of an issue when you're sending calls over a link with limited bandwidth. Usually more of a concern in the router. Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] AGI: Not getting answers from get_data in a call-file call
a) The call file Channel: Zap/g4/0409227633 MaxRetries: 0 RetryTime: 60 WaitTime: 30 Extension: 0409227633 Callerid: 0409227633 Context: barnet-callback Priority: 1 b) the snippet of extensions.conf that this is called in ; ; dial back ; exten = 0293353699,1,AGI(callback1.agi) [barnet-callback] exten = _.X,1,NoOp(callback time) exten = _.X,n,Answer() exten = _.X,n,AGI(callback2.agi) exten = _.X,n,Hangup exten = OutgoingSpoolFailed,1,NoOp(Failed) You dial in on 02 9335 3699, come into the AGI script which creates the above call-file (that works), it calls you back on via the call-file and then drops into the callback2.agi script which is the one I had in the earlier email. As I said earlier: nothing strange, nothing spectacular. Just strange that it doesn't work for calls initiated by Asterisk. -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ -- 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] AGI: Not getting answers from get_data in a call-file call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Have you tried with AGI Debug on? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHpoaqDQNt8rg0Kp4RAmDDAJ9DiVusLAV2ZlU42YKDr5uo05GIWgCeOliN 7Oe1nB1xymr3Uw+PlshOg3Y= =H5n9 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Edwin Groothuis wrote: I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); $AGI-answer(); my $i; $i = $AGI-channel_status(); $AGI-say_digits($i); $i = $AGI-get_data(one-moment-please, 1, 3); $AGI-say_digits($i); As you can see, nothing serious. When running this script in a normal telephone call, it works. When running this script in the call created with the call-file, I do hear the output of the first say_digits and the one-moment-please, but the pressing of the DTMF keys is not recognized by the system, the get_data() times out and the function returns nothing. This script once worked in 1.2.x (It was part of a bigger project) but now that I want to move it to 1.4 it gives this strange behaviour. Is there anybody with a hint on how to resolve this? You'd need to show us: a) The call file b) the snippet of extensions.conf that this is called in - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHpoHKDQNt8rg0Kp4RAskFAJ9T/K8gpl6HKa6mjRRNdm/r1MESGwCbBC4i R0buf80sRINRhdUGZk0asPY= =CmpE -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call
Have you tried with AGI Debug on? Yes! Even before you asked :-) This is when I use DeadAgi (for some reason): -- Executing [EMAIL PROTECTED]:3] DeadAGI(Zap/4:103-1, callback2.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/callback2.agi AGI Tx agi_request: callback2.agi AGI Tx agi_channel: Zap/4:103-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1202098745.49994 AGI Tx agi_callerid: 0288159096 AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 88159305 AGI Tx agi_rdnis: unknown AGI Tx agi_context: barnet-callback AGI Tx agi_extension: h AGI Tx agi_priority: 3 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx CHANNEL STATUS AGI Tx 200 result=6 AGI Rx SAY DIGITS 6 [Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to write frame -- Zap/4:103-1 Playing 'digits/6' (language 'en') AGI Tx 200 result=-1 AGI Rx GET DATA one-moment-please 1 3 [Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to write frame -- Zap/4:103-1 Playing 'one-moment-please' (language 'en') AGI Tx 200 result=-1 AGI Rx SAY DIGITS 1 [Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to write frame -- Zap/4:103-1 Playing 'digits/1' (language 'en') AGI Tx 200 result=-1 -- AGI Script callback2.agi completed, returning -1 And this is with normal AGI: -- Executing [EMAIL PROTECTED]:3] AGI(Zap/4:100-1, callback2.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/callback2.agi AGI Tx agi_request: callback2.agi AGI Tx agi_channel: Zap/4:100-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1202099898.50183 AGI Tx agi_callerid: 0288159096 AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 82572599 AGI Tx agi_rdnis: unknown AGI Tx agi_context: barnet-callback AGI Tx agi_extension: 0288159096 AGI Tx agi_priority: 3 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx CHANNEL STATUS AGI Tx 200 result=6 AGI Rx SAY DIGITS 6 -- Zap/4:100-1 Playing 'digits/6' (language 'en') AGI Tx 200 result=0 AGI Rx GET DATA one-moment-please 1 3 -- Zap/4:100-1 Playing 'one-moment-please' (language 'en') AGI Tx 200 result= (timeout) -- AGI Script callback2.agi completed, returning 0 When running it as a normal call (i.e. not initated by a call-file), it shows up with: AGI Rx GET DATA one-moment-please 1 3 -- Zap/93-1 Playing 'one-moment-please' (language 'en') AGI Tx 200 result=354 AGI Rx SAY DIGITS 354 -- Zap/93-1 Playing 'digits/3' (language 'en') -- Zap/93-1 Playing 'digits/5' (language 'en') -- Zap/93-1 Playing 'digits/4' (language 'en') AGI Tx 200 result=0 -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ -- 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] Telco MWI Detection on TDM400 Interface?
The MWI detection is done using fsk modem detection within chan_zap itself. (It does not support neon MWI detection.) The driver plays no real part in the detection. Doug Bailey - Original Message - From: Jim Duda [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 3, 2008 11:09:22 AM (GMT-0600) America/Chicago Subject: [asterisk-users] Telco MWI Detection on TDM400 Interface? I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming channel = 4 However, the zapnotify.sh script in the /usr/local/sbin directory is never getting called. Do I need a new version of the zaptel drivers? I'm currently using version 1.4.6 Is there anyway to get debugging into the log files? Am I missing something? Jim ___ -- 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] switch QOS requirements
Al lists wrote: Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. I guess to be safe, you would need to create 2 VLANS and in the switch define on VLAN as a high priority VLAN. At least for quite few years (more than 5), layer 2 switches from Nortel (disclaimer, I used to work for NT), would be able to match DSCP (or remark DSCP also, based in l3/l4 information) and give priority as defined by the user, to specific DSCP #, like EF to highest priority and goes on. I've no reason to believe other vendors don't have at least this capability. On Feb 3, 2008 7:06 PM, Michael Graves [EMAIL PROTECTED] wrote: On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote: John Williams [EMAIL PROTECTED] writes: We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks If you have one switch for the whole network, you're generally fine without QoS. Switches these days can handle full bandwidth on all ports at the same time. Anyway, Asterisk is no different from other PBX's when it comes to QoS. Should it turn out that you actually need it on the LAN, just be sure you set the tos parameters in sip.conf to something that is prioritized by the switch. It tends to be more of an issue when you're sending calls over a link with limited bandwidth. Usually more of a concern in the router. Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] switch QOS requirements
Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. I guess to be safe, you would need to create 2 VLANS and in the switch define on VLAN as a high priority VLAN. On Feb 3, 2008 7:06 PM, Michael Graves [EMAIL PROTECTED] wrote: On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote: John Williams [EMAIL PROTECTED] writes: We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks If you have one switch for the whole network, you're generally fine without QoS. Switches these days can handle full bandwidth on all ports at the same time. Anyway, Asterisk is no different from other PBX's when it comes to QoS. Should it turn out that you actually need it on the LAN, just be sure you set the tos parameters in sip.conf to something that is prioritized by the switch. It tends to be more of an issue when you're sending calls over a link with limited bandwidth. Usually more of a concern in the router. Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] Wait in Queue for 120 seconds for agent A to become free, THEN ring next agent
Hi all Just trying to set up a queue and wondering if this is possible. We have 3 agents, One of them is sort of the first point of contact What i am looking to do is 1. Someone rings the queue. 2. It rings Agent A.. If Agent A is on the phone then put them on hold for 120 seconds, and if Agent A gets off the phone within those 120 seconds, put the call to them. 3. If 120 seconds expires, then call agent B, if B rings out, Call agent C So all i need is for the call to wait 120 seconds for agent A to get off the phone. Then progress the call if they dont. Is this possible? If so, any pointers? Cheers - Kev -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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 with DTMF dialing
Hi Thanks for the response Anthony Messina said the following on 01-Feb-08 03:36 PM: On Thursday 31 January 2008 11:52:09 pm Ian wrote: Sorry for taking so long to reply, This email got lost in translation, again. Ian Ian said the following on 30-Jan-08 03:57 PM Thaks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 did you use zaptel-1.4.7 prior to this? did it work then? if so, it may be related to http://bugs.digium.com/view.php?id=11855 No, this is a clean install, I will download 1.4.7 tonight and recompile. Any specific things I should watch for, like would I need to recompile Asterisk when I compile Zaptel, etc. Thanks for the link, I am leaving a comment there as well. Regards Ian ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.18/1255 - Release Date: 2/1/2008 9:59 AM ___ -- 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 version to be downloaded for my machines
Здравствуйте, Mindaugas. MK Download for Pentium4 MK The output of cat /proc/cpuinfo giving a [Intel (R) MK Pentium (R) D] so what is the g729 version I have to MK download to work with my machine? If it is not loaded in Asterisk successful - try for Pentium -- Alexey mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with IRQ Share
Hi I have a Server with Centos 5, TDM400p, HP Server ML110. My problem is that I see IRQ Share with my TDM400P. How can I fix that??? ___ -- 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] switch QOS requirements
Al lists [EMAIL PROTECTED] writes: Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. Sadly Asterisk still calls DSCP TOS, so I stuck with Asterisk nomenclature. Modern switches can translate DSCP into COS (lossily) and many do so automatically. With most switches, sending DSCP set to EF will give the traffic priority. Signalling (SIP) is usually set to class AF21; Cisco recommends that CS3 is used instead. Either should work fine with the default setup on many LAN switches. I guess to be safe, you would need to create 2 VLANS and in the switch define on VLAN as a high priority VLAN. That certainly works too. It's just a pain if you use the built-in switches available in most phones. You need to manually tell them to use a different VLAN, or use Cisco-switches with CDP. Most phone manufacturers don't support LLDP for voice VLAN discovery yet. /Benny ___ -- 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] transcoder
Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 . and forward it to a media gateway .. Regards Khaled chehab * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 with DTMF dialing
Thanks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost routing. The problem I am having is dialing out using DTMF signalling. At the moment I am making do with Pulse dialing through the 3 analog lines. I can recieve calls on the Cellphone line without any problems, but cant dial out through it, as a cellphone cant do pulse dialing. I have run ztmonitor 1 -f gains, where 1 is the zap channel where the cellphone is located, while dialing the number 072 031 1294. I then went to audacity, on my own pc, and converted the raw file into mp3 format, mp3 is a compressed format, and hence may lose some quality. Generally you should stick with wav. ztmonitor should spit the appropriate sox command to do the conversion. Maybe it would look slightly different in the original format. Ok I tried this everywhich way I could but everytime I came up short of an answer. Meaning I am unable to find the right sox command to get this converted to wav on the same computer, so once again I got it to my pc, and then using my favourite friend, audacity I imported it as a raw format at 8000Hz, and exported it as a wav file this time, available for download from http://www.iancoetzee.za.net/gain.wav. it has the same effect, the numbers I dialed and the feedback I got is two different things. which is available for download at http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the playback I concluded that the DTMF signals being sent is totally wrong. Is that the whole tone? It is too short to be a valid DTMF. Yes that was the dial bit, this time I included the whole recording from beginning to end. if you count the tones you get to 10, which is the correct amount for South Africa. Another thing that got me worried is the fact that the last digit has a fair ammount of pause (about the same length of another tone) before it is sent. If you want I can upload the raw data to my server as well. Regards Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Technician Telephone : 012 664 2300 cellphone : 079 522 6519 Fax : 012 644 2902 E-mail : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- 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
