[asterisk-users] Extend time in call pickup
Hi, could anyone tell me how to extend the time for Asterisk to pick up an incoming PSTN call ? Thanks Gidean___ --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
[asterisk-users] Can zaptel freak out if you configure 2 trunks but use only one?
I am using Asterisk 1.2.13 with Zaptel 1.2.11, I used to have an old PBX connected to one port and the PRI connected to the other. I'm having serious stability issues with Asterisk on a box that has been rock solid previously. The old PBX died two months ago so one port on the TE210P is now unused but still configured. Also I'm afraid I have upgraded from Asterisk 1.2.9.1 and the old zaptel version because of the security flaws. I'm now puzzled why Asterisk is being unstable. I do a nightly restart because Asterisk is extremely slow in trying to resolve failed dns lookups for providers. Often Asterisk will keep running or restart properly but on the console I can see it restarting / restarting the B-channels really slow (normally to restart all B channels takes max 1 second) i can really see it restarting one channel in about one second, and some channels are skipped. It also happens that Asterisk refuses to start at all. A reboot seems the only solution to the problem. Could it be that the configured (but unused) trunk is causing me problems, buffer overruns or anything similar? Or is this an issue with more recent zaptel and asterisk configs? Thanks all!! ___ --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
Re: [asterisk-users] bristuff error: received SETUP message for callthat is not a new call
On Mon, Dec 04, 2006 at 08:13:43AM +0100, Koopmann, Jan-Peter wrote: On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote: Hello, With the following setup: - asterisk 1.2.13, - zaptel 1.2.10 - bristuff 0.3.0-PRE-1v - quadbri card, Have you tried using bristuff 1v with the qozap driver of 1s? All qozap versions after 1s had serious problems (which seem to be fixed in soon to be released 1w). If this does not help, do a pri debug or better yet pri intense debug, describe the problem and contact the author with this info. I have just written: http://www.voip-info.org/wiki/view/Bristuff Please post issues there. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Is there any Asterisk controllable thermostat?
Asterisk can control any x10 capable device. For a good example, see http://lorancestinson.blogspot.com/2006/08/asterisk-can-control-world.ht ml From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Sunday, December 03, 2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is there any Asterisk controllable thermostat? I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --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
Re: [asterisk-users] TDM01B installation
On Mon, Dec 04, 2006 at 02:58:29PM +0700, Mochamad Susantok wrote: On Mon, Dec 04, 2006 at 09:12:21AM +0700, Mochamad Susantok wrote: Hi, iam new in this milis I have problem with TDM01B Installation, output zttool command is Unable to open /dev/zap/ctl: No such device or address and then i find the same IRQ uses VGA compatible controller and Communication controller is 169 What can i do next ? please your advice cat /proc/zaptel/* No such file or directory lspci Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface modinfo | grep wctdm nothing ouptut What version of Zaptel do you use? What distribution? Debian 2.4 kernel asterisk-1.0.7 zaptel-1.0.7 That version of Zaptel is probably too old and does not support latest TDM400P cards. Consider using zaptel debs from deb http://updates.xorcom.com/rapid sarge main -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOW TO - Asterisk apps/modify and compile
On Mon, Dec 04, 2006 at 12:41:46PM +0530, Thirumal Saminathan wrote: hi all, i need to integrate and modify one of the application in asterisk/apps section... whenever i modified small steps ..in order to check and compile i 've to do recompile the whole asterisk module and it consuke to much time... please anyone couls you tell me, how can i modify it , compile and test the I/O in asterisk applications in a easy way... When you re-run 'make' on a built tree, it will pass on the whole tree, but will only actually build the changed apps. Instead of running a complete install, it may be simpler to manually copy the modified app to /usr/lib/asteris/modules . You should probably be in better place to decide when to backup a copy and when to delete. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 200+ analog phones connected to FXS modules
CG == Csibra Gergo [EMAIL PROTECTED] writes: CG Well, the data bandwidth is only one. The irq is the other, and CG that is the bottleneck. You get 1000 interrupts per second. If that's a bottleneck then there's something fundamentally wrong with your system. /Benny ___ --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
[asterisk-users] Nokia E60 problems
Hi, I am testing Nokia E60 with Asterisk. And I noticed that if another side is busy, nokia is still calling (I hear alerting), it do not show that another side is busy. Maybe somebody has noticed the same problem too adnd solved this one. I made the same tests with Xlite and don't have any problems like nokia. Please help me ___ --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
[asterisk-users] Problem with h323 support
Hi, I have Asterisk SVN-branch-1.4-r47845 installed in a Ubuntu Dapper. Its works as sip server and I am trying to get h323 support. I installed these packages: libpt-1.10.0 libopenh323-1.18.0 And I set the next global variables: PWLIBDIR=/usr/share/pwlib OPENH323DIR=/usr/share/openh323 Then, when I execute the configure script (before installation) and finishes with this message: # ./configure --with-h323 . . . checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/share/pwlib/make/ptlib-config checking if PWLib version 1.10.0 is compatible with chan_h323... yes checking PWLib installation validity... no configure: *** configure: *** The OPENH323 installation on this system appears to be broken. configure: *** including --without-h323 What is the problem in my configuration? Regards, Diego. ___ --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
Re: [asterisk-users] Problem with h323 support
On Mon, Dec 04, 2006 at 01:12:02PM +0100, Diego Moreno wrote: Hi, I have Asterisk SVN-branch-1.4-r47845 installed in a Ubuntu Dapper. Its works as sip server and I am trying to get h323 support. I installed these packages: libpt-1.10.0 libopenh323-1.18.0 And I set the next global variables: PWLIBDIR=/usr/share/pwlib OPENH323DIR=/usr/share/openh323 Then, when I execute the configure script (before installation) and finishes with this message: # ./configure --with-h323 . . . checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/share/pwlib/make/ptlib-config checking if PWLib version 1.10.0 is compatible with chan_h323... yes checking PWLib installation validity... no configure: *** configure: *** The OPENH323 installation on this system appears to be broken. configure: *** including --without-h323 What is the problem in my configuration? This is a problem with the assumptions that the configure script makes regarding the scructture of the h323 tree. After some mucking with configure.ac I managed to get this working, but I can't seem to find my patch anywhere. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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Re: [asterisk-users] mwi for voicemail not showing up for realtime config.
Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --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
RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Hi Scott, I have the following firmware 1.1.0.16 1.1.0.11 1.1.1.9 1.1.1.14 1.1.2.6 1.1.2.13 Some of these were not from the official website but they were all an improvement 1.1.2.13 is very stable apart from the 56 button ext, unit support. Let me know which ones you want and I can send them to you. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --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 ___ --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
Re: [asterisk-users] MySQL cmd % pattern matching
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Garth van Sittert [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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Re: [asterisk-users] MySQL cmd % pattern matching
Garth van Sittert wrote:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
This is how I would do it:
exten = s,5,MYSQL(Query resultid ${connid} SELECT name FROM contacts
WHERE tel = ${number})
exten = s,6,MYSQL(Fetch fetchid ${resultid} contact.name)
exten = s,7,MYSQL(Disconnect ${connid})
exten = s,8,MYSQL(Clear ${resultid})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Henry is my newest Hero :) I'll coordinate with you directly on the releases. Thank you. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry.L.Coleman Sent: Monday, December 04, 2006 4:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have the following firmware 1.1.0.16 1.1.0.11 1.1.1.9 1.1.1.14 1.1.2.6 1.1.2.13 Some of these were not from the official website but they were all an improvement 1.1.2.13 is very stable apart from the 56 button ext, unit support. Let me know which ones you want and I can send them to you. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --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 ___ --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
[asterisk-users] forward skinny call to SIP
Hi i have to do the following setup: 1 - i receive a call on skinny protocol 2 - i forward the call to a sip user I think that the skinny phone must be registered on asterisk, in a particular extension, for example: [skinny_internat_ext] 987,1,Dial(SIP/[EMAIL PROTECTED]) And if the skinny phone dials 987 i make the call to SIP/user. But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --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 ___ --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
Re: [asterisk-users] forward skinny call to SIP
I think, you can't make skinny call without phone registered to any call control server. If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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
Re: [asterisk-users] forward skinny call to SIP
If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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
RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363highlight= Regards, Scott -Original Message- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --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 ___ --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
Re: [asterisk-users] forward skinny call to SIP
i callmanager add asterisk as h323 gateway and also add route pattern to this gateway compile asterisk with h323 support, it will build chan_h323.so, add callmanager as friend in h323.conf in callmanager v4 you can also use SIP trunk between callmanager and asterisk, but keep in mind, that only g711 codec is supported on SIP trunk. PJ nik600 wrote: If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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
Re: [asterisk-users] MySQL cmd % pattern matching
Hi Jon
No luck - it works with the quotes and no % sign but as soon as I add
the % it doesn't work.
Garth
Jon Farmer wrote:
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Garth van Sittert [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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Re: [asterisk-users] MySQL cmd % pattern matching
I have it working as your example, Doug, but unfortunately I need the
like phrase as the numbers all contain spaces or sometimes even brackets.
Garth
Doug Lytle wrote:
Garth van Sittert wrote:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
This is how I would do it:
exten = s,5,MYSQL(Query resultid ${connid} SELECT name FROM contacts
WHERE tel = ${number})
exten = s,6,MYSQL(Fetch fetchid ${resultid} contact.name)
exten = s,7,MYSQL(Disconnect ${connid})
exten = s,8,MYSQL(Clear ${resultid})
Doug
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Re: [asterisk-users] MySQL cmd % pattern matching
Ah, I found the solution to my problem:
I slashed out the ' to give:
exten = s,5,Set(query=SELECT name from contacts where tel like
\'%${number}\')
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
Kind Regards
Garth
Garth van Sittert wrote:
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and %
in the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how
to hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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Re: [asterisk-users] MySQL cmd % pattern matching
Try prefixing the % with a \.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 4 Dec 2006, Garth van Sittert wrote:
Hi Jon
No luck - it works with the quotes and no % sign but as soon as I add
the % it doesn't work.
Garth
Jon Farmer wrote:
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Garth van Sittert [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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[asterisk-users] Codec transcoding and call recording
heres my scenariosoftphone-Asterisk( outgoing call recording )Call Provider I am recording all outgoing calls on asterisk so its obvious that there is no native bridging . Suppose if i am using gsm from softphone--asterisk and then what codec should i prefer for asterisk- provider ?? Bandwith is not at all a problem between asterisk-provider connection . So if i am using gsm from softphone to asterisk ... then what would be better choice ? sending gsm to provider or ulaw/alaw ? I suppose asterisk already touches audio stream since it is recording to wav file but what i want to know is do asterisk sends same gsm stream ahead to provider ( and reduces cpu usage ) or it still have to transcode for recording ? because if its transcoding anyway then sending ulaw/alaw stream ahead would be better right ? or am i missing something :-/ Also what would be better if i use g729 from softphone--asterisk( capable of g729 transcoding ) should i send g729 ahead or ulaw/alaw? ___ --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
Re: [asterisk-users] Problem with h323 support
Thanks for your answer, Tzafrir. I installed the package libopenh323-1.18.0 with the Synaptic application. Then, I suppose the structure of the h323 tree is correct. But I don't know if the problem is in my system (installation and configuration) or in the configure script for the asterisk installation. What do you think? Regards, Diego. 2006/12/4, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Dec 04, 2006 at 01:12:02PM +0100, Diego Moreno wrote: Hi, I have Asterisk SVN-branch-1.4-r47845 installed in a Ubuntu Dapper. Its works as sip server and I am trying to get h323 support. I installed these packages: libpt-1.10.0 libopenh323-1.18.0 And I set the next global variables: PWLIBDIR=/usr/share/pwlib OPENH323DIR=/usr/share/openh323 Then, when I execute the configure script (before installation) and finishes with this message: # ./configure --with-h323 . . . checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/share/pwlib/make/ptlib-config checking if PWLib version 1.10.0 is compatible with chan_h323... yes checking PWLib installation validity... no configure: *** configure: *** The OPENH323 installation on this system appears to be broken. configure: *** including --without-h323 What is the problem in my configuration? This is a problem with the assumptions that the configure script makes regarding the scructture of the h323 tree. After some mucking with configure.ac I managed to get this working, but I can't seem to find my patch anywhere. -- 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 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
[asterisk-users] Moderate setup
I am planning to put up a asterisk server with around 50-60 phones over a lan . I am planning on keeping a decent server ( for outbound pstn ) and all phones connected via linksys pap2 ( all 60 phones as pap2 registering to asterisk) . Does this kind of setup give problem ? ___ --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
Re: [asterisk-users] Re: 200+ analog phones connected to FXS modules
On Saturday 02 December 2006 14:26, Csibra Gergo wrote: Well, the data bandwidth is only one. The irq is the other, and that is the bottleneck. Unless I am mistaken (first time for everything, g), the mutli-span cards issue 1 interrupt per every millisecond, and *all* spans on that card are serviced in that interrupt. To put it another way: TDM4xxP - 1-4 channels, 1000Hz interrupt rate TE100P - 1-24 channels, 1000Hz interrupt rate TE2xxP - 1-48 channels, 1000Hz interrupt rate TE4xxP - 1-96 channels, 1000Hz interrupt rate I know that multiple cards do not share the zaptel interrupt, so with two cards you have two 1000Hz interrupt sources. I have a back-burner I wonder if kind of idea which involves zaptel shutting down the interrupt on the 2nd (and more) cards and servicing all cards in one interrupt, but the back burner's so far away now I don't know if I'll ever get to it. :-) -A. ___ --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
Re: [asterisk-users] Digium through Octasic
On Sunday 03 December 2006 03:17, Julian Lyndon-Smith wrote: Is there a trade-in program in place ? I have a te410p and a te405p that I am not using because of various problems we had, but would like to try the te407 ... [EMAIL PROTECTED] can surely help you here... -A. ___ --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
Re: [asterisk-users] Re: 200+ analog phones connected to FXS modules
Monday, December 4, 2006, 12:18:16 PM, Benny Amorsen wrote: CG == Csibra Gergo [EMAIL PROTECTED] writes: CG Well, the data bandwidth is only one. The irq is the other, and CG that is the bottleneck. You get 1000 interrupts per second. If that's a bottleneck then there's something fundamentally wrong with your system. Well, I think there's far more htan 1000 interrupts come from an T1/E1 card. Or do you think 1000/channel? -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --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
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
On Mon, 2006-12-04 at 00:58 -0700, [EMAIL PROTECTED] wrote: Date: Sun, 3 Dec 2006 23:04:52 -0500 From: Zeeshan Zakaria [EMAIL PROTECTED] Subject: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I am wondering if there is any such thermostat available which can be controlled from Asterisk. Trixbox comes bundled with xPl, which is a home automation network API that is also common to Windows XP. I haven't seen any documentation of how to actually use it (with Trixbox/Asterisk), but I would be very interested in seeing some, including examples and supported HW. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria -- (C) Matthew Rubenstein ___ --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
[asterisk-users] google talk
hi How does asterisk can act as google talk's client. for mapping, received calls , to google talk. tanx Mani Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com ___ --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
[asterisk-users] Odd queue issue
Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2--A---IAX---B---SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP---A---IAX---B---SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to server A dials the 800 number I have it go directly to server B with: extensions_custom.conf:exten = 1866611,1,Dial(IAX2/callcenter/866611) This is what I get in the logs for the failed call: Dec 4 10:01:41 VERBOSE[14045] logger.c: -- Called 126 Dec 4 10:01:41 VERBOSE[14008] logger.c: -- Agent/ is ringing Dec 4 10:01:43 DEBUG[15071] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Dec 4 10:01:48 DEBUG[15071] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Dec 4 10:01:48 WARNING[15071] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) Dec 4 10:01:48 WARNING[15071] chan_sip.c: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Dec 4 10:01:48 DEBUG[14045] chan_sip.c: update_call_counter(126) - decrement call limit counter Dec 4 10:01:48 VERBOSE[14045] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 4 10:01:48 DEBUG[14008] app_queue.c: Dunno what to do with control type -1 Dec 4 10:01:48 DEBUG[14045] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Dec 4 10:01:48 DEBUG[14045] pbx.c: Expression result is '1' Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Executing GotoIf(Local/[EMAIL PROTECTED],2, 1?s-CHANUNAVAIL|1) in new stack Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Goto (macro-exten-vm,s-CHANUNAVAIL,1) Dec 4 10:01:48 VERBOSE[14045] logger.c: -- Executing Congestion(Local/[EMAIL PROTECTED],2, ) in new stack Dec 4 10:01:48 VERBOSE[14008] logger.c: -- Agent/ is circuit-busy Dec 4 10:01:48 DEBUG[14008] chan_agent.c: Hangup called for state Down Dec 4 10:01:48 DEBUG[14008] chan_agent.c: Hungup, howlong is 7, autologoff is 28 Dec 4 10:01:48 DEBUG[14008] app_queue.c: Everyone is busy at this time Dec 4 10:01:48 VERBOSE[14045] logger.c: == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'exten-vm' Extention 126 is indeed my phone... and my agent ID is . If I call in from the IAX2 provider from outside the system I get: Dec 4 10:06:43 VERBOSE[14214] logger.c: -- Called 126 Dec 4 10:06:43 VERBOSE[14197] logger.c: -- Agent/ is ringing Dec 4 10:06:43 DEBUG[15071] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 4 10:06:43 DEBUG[15071] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 4 10:06:43 VERBOSE[14214] logger.c: -- SIP/126-7d1b is ringing Dec 4 10:06:47 VERBOSE[15071] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Dec 4 10:06:47 DEBUG[15071] channel.c: Scheduling timer at 160 sample intervals Dec 4 10:06:47 DEBUG[14188] channel.c: Generator got voice, switching to phase locked mode Dec 4 10:06:47 DEBUG[14188] channel.c: Scheduling timer at 0 sample intervals Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1228778601: Match Found Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Acked pending invite 102 Dec 4 10:06:47 DEBUG[15071] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Dec 4 10:06:47 DEBUG[15071] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Dec 4 10:06:47 DEBUG[15059] channel.c: Avoiding initial deadlock for 'SIP/126-7d1b' Dec 4 10:06:47 VERBOSE[14214] logger.c: -- SIP/126-7d1b answered Local/[EMAIL PROTECTED],2 Dec 4 10:06:47 DEBUG[14197] app_queue.c: Dunno what to do with control type -1 Dec 4 10:06:47 VERBOSE[14197] logger.c: -- Agent/ answered IAX2/serverA-1 Dec 4 10:06:47 DEBUG[15059] channel.c: Avoiding initial deadlock for 'Local/[EMAIL PROTECTED],2' Dec 4 10:06:47 DEBUG[14197] channel.c: Scheduling timer at 160 sample intervals Dec 4 10:06:47 VERBOSE[14197] logger.c: -- Playing 'custom/CountDown' (language 'en') Dec 4 10:06:48 DEBUG[14214] channel.c: Planning to masquerade channel SIP/126-7d1b into the structure of Local/[EMAIL PROTECTED],1 Dec 4 10:06:48 DEBUG[14214] channel.c: Done planning to masquerade channel SIP/126-7d1b into the structure of Local/[EMAIL PROTECTED],1 Dec 4 10:06:48 DEBUG[14214] chan_local.c: Not posting to queue since already masked on 'Local/[EMAIL PROTECTED],2' Dec 4 10:06:48 DEBUG[14197] channel.c: Got clone lock for masquerade on
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I would really like to see some documentation also. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Matthew Rubenstein wrote: On Mon, 2006-12-04 at 00:58 -0700, [EMAIL PROTECTED] wrote: Date: Sun, 3 Dec 2006 23:04:52 -0500 From: Zeeshan Zakaria [EMAIL PROTECTED] Subject: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I am wondering if there is any such thermostat available which can be controlled from Asterisk. Trixbox comes bundled with xPl, which is a home automation network API that is also common to Windows XP. I haven't seen any documentation of how to actually use it (with Trixbox/Asterisk), but I would be very interested in seeing some, including examples and supported HW. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? -- Zeeshan A Zakaria ___ --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
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
On Sun, Dec 03, 2006 at 11:04:52PM -0500, Zeeshan Zakaria wrote: I am wondering if there is any such thermostat available which can be controlled from Asterisk. Like you call your home pbx, dial some extension, e.g. 333 and it asks to set the temperature, you enter a temperature, and it sets the thermostat to that temperature. This thermostat will be very useful, e.g. when you're coming back home after a few days and now its snowing and you want home to be warm on your arrival, you can turn the furnace on an hour before your arrival. Is there any such thermostat available, and for that matter any other Asterisk controllable home automation devices? The first question you should ask yourself is: can Linux [or any other specific OS you run Asterisk on] contro a thermostat. Once you managed to do that, connecting it to Asterisk shouldn't be too big a deal. If all else fails, use AGI for quickdirty patching with external scripts. -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No answer when press 0 for operator in VM in 1.0 .9?
Users cannot dial 0 to get to the operator in voicemail. * 1.0.9 Linux asterisk1.local 2.4.21-32.EL #1 Wed May 18 18:31:54 EDT 2005 i686 athlon i386 GNU/Linux (CentOS) Snom 360 DTMF=RFC2833 Switched LAN, no problems w/ DTMF anywhere operator=yes in voicemail.conf this does not apply to my situation: http://bugs.digium.com/bug_view_page.php?bug_id=0003080 When OGM is played, you press 0, nothing on the console. When it starts recording, you can press Zero-Zero and this is what happens: -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2, 1?3:4) in new stack -- Goto (dial-internal,ivr-vm,3) -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/2, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/0552/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/0552/INBOX/msg format: wav49, 0x8694d40 -- x=1, open writing: /var/spool/asterisk/voicemail/default/0552/INBOX/msg format: wav, 0x8654d88 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') -- Playing 'vm-deleted' (language 'en') -- Playing 'transfer' (language 'en') -- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, INBOUND Regular call exiting for user Hansen Li from Colin Anderson 7028247) in new stack -- Hungup 'IAX2/[EMAIL PROTECTED]/2' Any pointers would be welcome. ___ --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
Re: [asterisk-users] forward skinny call to SIP
thanks can you explain me how to compile asterisk with h323 support? or is it biult in by default? On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: i callmanager add asterisk as h323 gateway and also add route pattern to this gateway compile asterisk with h323 support, it will build chan_h323.so, add callmanager as friend in h323.conf in callmanager v4 you can also use SIP trunk between callmanager and asterisk, but keep in mind, that only g711 codec is supported on SIP trunk. PJ nik600 wrote: If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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 ___ --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
RE: [asterisk-users] Help with IAX Trunk
Yes. That was the solution. Not sure why that 'r' is there in the first place David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Saturday, December 02, 2006 11:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with IAX Trunk On 09:48, Sat 02 Dec 06, Dave Morrow wrote: H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum =2 Probably yeah. The r option in the dial command will not pass early media but instead generates it's own. I find the r flag for dial and queue the wrong thing to do. In dial it will disable stuff like 'this call will cost you 300 euro a minute and that's something I really wanna hear. In queue() it will kill the periodic announcements. annoying as well. I removed them from everywhere in my extensions.conf and my system is much more usable. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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 ___ --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
Re: [asterisk-users] Moderate setup
Vicky wrote: I am planning to put up a asterisk server with around 50-60 phones over a lan . I am planning on keeping a decent server ( for outbound pstn ) and all phones connected via linksys pap2 ( all 60 phones as pap2 registering to asterisk) . Does this kind of setup give problem ? I would go with a few Quintum TenorAX boxes, especially if you already have CAT3 cable running to your workstations. Thanks, Steve ___ --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
Re: [asterisk-users] forward skinny call to SIP
1.4 - make menuselect 1.2 - make in channels/h323 (read readme.txt here) nik600 wrote: thanks can you explain me how to compile asterisk with h323 support? or is it biult in by default? On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: i callmanager add asterisk as h323 gateway and also add route pattern to this gateway compile asterisk with h323 support, it will build chan_h323.so, add callmanager as friend in h323.conf in callmanager v4 you can also use SIP trunk between callmanager and asterisk, but keep in mind, that only g711 codec is supported on SIP trunk. PJ nik600 wrote: If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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 ___ --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
[asterisk-users] Addqueuemember and roaming users problem.
Hi,
I'm having hard time to emulate agencallbacklogin. Agent can logon
and receive call without any problem using addqueuemember. The problem comes
when I try to evaluate their performance using queuemetrics. Here is an
exemple of my log script:
;Agent Login
exten = _60XXX,1,Macro(agentLogin)
[macro-agentlogin]
exten = standard,1,AddQueueMember(queue1) exten =
standard,n,AddQueueMember(queue2) exten = standard,n,AddQueueMember(queue3)
exten = standard,n,AddQueueMember(queue4) exten = standard,n,System(echo
${EPOCH}|${UNIQUEID}|NONE|SIP/${MACRO_EXTEN:2}|AGENTLOGIN|-
/var/log/asterisk/queue_log) exten = standard,n,Wait(0.5) exten =
standard,n,Playback(agent-loginok)
exten = standard,n,Hangup()
In queuemetrics realtime panel, I can see the name of the agent who is
logged because of this part : SIP/${MACRO_EXTEN:2}
But, when Asterisk sends a call to the agent, he sends the call to the
station where the agent is sit and not to the agent himself. I explain: if
agent 204 is sit a desk 260 then in the realtime panel I can see that agent
204 is on but if I analyse the report, this agent didn't received any call,
it's 260 that answered the call. So if more than one person use this
station, I cannot know how many call as been taken by every agent.
With agentcallback login, this wasn't a problem because of the use of way it
handle logon and login.
Anyone has been able to fix this problem ?
David
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Re: [asterisk-users] forward skinny call to SIP
i am trying to download Open H.323 version v1.17.1, PWLib v1.9.0 but http://www.openh323.org/ seems to be down, can you suggest me an alternative link where to download them? many thanks.. On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: 1.4 - make menuselect 1.2 - make in channels/h323 (read readme.txt here) nik600 wrote: thanks can you explain me how to compile asterisk with h323 support? or is it biult in by default? On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: i callmanager add asterisk as h323 gateway and also add route pattern to this gateway compile asterisk with h323 support, it will build chan_h323.so, add callmanager as friend in h323.conf in callmanager v4 you can also use SIP trunk between callmanager and asterisk, but keep in mind, that only g711 codec is supported on SIP trunk. PJ nik600 wrote: If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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 ___ --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 ___ --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
Re: [asterisk-users] forward skinny call to SIP
I would recommend using SIP instead of h323, if at all possible. - nik600 [EMAIL PROTECTED] wrote: i am trying to download Open H.323 version v1.17.1, PWLib v1.9.0 but http://www.openh323.org/ seems to be down, can you suggest me an alternative link where to download them? many thanks.. On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: 1.4 - make menuselect 1.2 - make in channels/h323 (read readme.txt here) nik600 wrote: thanks can you explain me how to compile asterisk with h323 support? or is it biult in by default? On 12/4/06, Pavel Jezek [EMAIL PROTECTED] wrote: i callmanager add asterisk as h323 gateway and also add route pattern to this gateway compile asterisk with h323 support, it will build chan_h323.so, add callmanager as friend in h323.conf in callmanager v4 you can also use SIP trunk between callmanager and asterisk, but keep in mind, that only g711 codec is supported on SIP trunk. PJ nik600 wrote: If you have skinny phone registered eg. in ci$co callmanager, you should make h323 trunk between asterisk and callmanager. PJ yes, this is my scenario, sorry for my bad explanation... How can i do that? nik600 wrote: But how can i do that if the skinny phone isn't registered to Asterisk? Thanks in advance ___ --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 ___ --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 ___ --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 ___ --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 -- Jason Parker Digium ___ --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
[asterisk-users] Sangoma a301 or other DS3 card
Hi, Is anyone here using any DS3 card (like Sangoma A301) in Asterisk, to handle 672 voice channels? If so, which hardware are you using? Which driver? Did you have any problem related to echo cancellation? -Lars ___ --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
Re: [asterisk-users] forward skinny call to SIP
yes, but as I said, callmanager v4 supports only g711 codecs over SIP trunk :-( if you have some phones in callmanager's region g729 (over WAN) and would like to call to asterisk from this phones, you need to use g729 on trunk, that is currently in callmanager possible only with h323. maybe this limitation is away in callmanager v5, I don't know. PJ Jason Parker wrote: I would recommend using SIP instead of h323, if at all possible. ___ --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
Re: [asterisk-users] forward skinny call to SIP
http://sourceforge.net/projects/openh323 nik600 wrote: i am trying to download Open H.323 version v1.17.1, PWLib v1.9.0 but http://www.openh323.org/ seems to be down, can you suggest me an alternative link where to download them? many thanks.. ___ --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
Re: [asterisk-users] AsteriskNow console access
me too, i'm trying to add sip users , i click save, it reports successfully saved... but there are no sip accounts created... On 11/29/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: i had the same problem. the GUI stopped responding to configuration changes. On 11/28/06, James Willing [EMAIL PROTECTED] wrote: Geoff Karl [EMAIL PROTECTED] wrote: I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? Yes, as I found out (rather painfully) after the second (or was it third) install, for console access you have to login as 'admin', using the password you entered during the installation. And I agree that it looks promising, though as far as I can tell so far none of the GUI functionality actually works yet. So far, I have been unable to actually get it to commit any changes entered via the GUI and after a few attempts (or an hour or so of running) the GUI generally appears to stop functioning. No response to 'system info' selection, etc... ...but it is 'Beta 1' afterall... B^} -- -jim (Willing) Midwest Connections, Inc. ___ --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 ___ --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
Re: [asterisk-users] forward skinny call to SIP
On Mon, Dec 04, 2006 at 05:32:26PM +0100, nik600 wrote: i am trying to download Open H.323 version v1.17.1, PWLib v1.9.0 but http://www.openh323.org/ seems to be down, can you suggest me an alternative link where to download them? openh323.org is unrelated to current developemnt of openh323. Try http://www.voxgratia.org/ -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Answer a call that is not ringing on your extension
Answer a call that is not ringing on your extension. I want to pick up an external call that is ringing on another extension that is not mine. Now in my old standard pbx I press 5 and I get the call. How to do this with asterisk? thank you. ___ --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
Re: [asterisk-users] Sangoma a301 or other DS3 card
On Mon, 2006-12-04 at 14:04 -0300, Lars Knopf wrote: Hi, Is anyone here using any DS3 card (like Sangoma A301) in Asterisk, to handle 672 voice channels? Afaik the Sangoma card is not channelized so can not be used with Asterisk. If so, which hardware are you using? Which driver? Did you have any problem related to echo cancellation? If you are going to use one or more DS3 links I think you are better off using a tad more carrier class stuff like the Lucent APX8000 or Cisco 5XXX boxes. Or check eBay for the MaxTNT which will work fine too. Regards, Patrick ___ --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
Re: [asterisk-users] Answer a call that is not ringing on your extension
David Parcerisa wrote: Answer a call that is not ringing on your extension. I want to pick up an external call that is ringing on another extension that is not mine. Now in my old standard pbx I press 5 and I get the call. How to do this with asterisk? thank you. ___ --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 Hi David You can use callgroup/pickupgroup in sip.conf, and use features.conf to choose what you have to press on the phones to pickup other calls. All phones have to be in the same group (0-64 I think). Regards, Ove ___ --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
[asterisk-users] ASterisk and SER
HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun ___ --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
Re: [asterisk-users] forward skinny call to SIP
Thanks to all as explained in channels/h323/README i am doing the following: cd /path/to/pwlib ./configure make clean opt PWLIBDIR=$HOME/pwlib export PWLIBDIR cd /path/to/openh323 ./configure make clean opt cd /path/to/asterisk/channels/h323 make opt cd /path/to/asterisk make install but when i try to compile openh323 i get this error: h323ep.cxx: In member function `PNatMethod* H323EndPoint::GetPreferedNatMethod(const PIPSocket::Address)': h323ep.cxx:3211: error: 'class PNatStrategy' has no member named 'GetNATList' h323ep.cxx:3215: error: 'class PNatMethod' has no member named 'GetNatMethodName' make[1]: *** [/home/programmi/asterisk-1.2.13_h323/openh323_v1_19_0_1/lib/obj_linux_x86_r/h323ep.o] Error 1 make[1]: Leaving directory `/home/programmi/asterisk-1.2.13_h323/openh323_v1_19_0_1/src' make: *** [opt] Error 2 the version i am compiling are: openh323-v1_19_0_1 pwlib-v1_10_2 an i wanto to install them on asterisk-1.2.13 ___ --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
[asterisk-users] Registering VoIP providers with realtime
Hi guys, I'm using realtime DB connection to get sip.conf and voicemail.conf data from a MySQL engine. I've been wondering if I could include my VoIP providers (the entry register = user:[EMAIL PROTECTED]/extension that is found in sip.conf) in the database so that a provider peer can be dynamically retrieved from the DB, but I had no luck at this time. Did any of us find the way to include VoIP providers in the realtime structure? Thanks in advance! ___ --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
Re: [asterisk-users] forward skinny call to SIP
ok, it seems 've resolved compiling stable version openh323_v1_18_0 pwlib_v1_10_0 ___ --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
[asterisk-users] Re: Recommendation for FXO
On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said: snip So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? I am wondering if anyone has experience with the Audiocodes MP114 (2fxs/2fxo)? This is pricey, but I am SO sick of mucking around with consumer grade bs (ie grandstream). I am also curious about the mediatrix 1204 and the Multitech MVP-130 although it sort of looks to me like the multitech doesn't do SIP. Any thoughts or help, before I take an expensive leap? Marty PS asterisk 1.2.13 ___ --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
Re: [asterisk-users] ASterisk and SER
On 04/12/06, Arun Kumar [EMAIL PROTECTED] wrote: HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. Your dialplan. (Since you didn't get around to posting any configuration or log information, that's about as close as anyone's going to get to your problem). Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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
RE : [asterisk-users] Re: Recommendation for FXO
Hi the list and Marty, Take a look to www.aliwei.com. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 4 décembre 2006 20:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: Recommendation for FXO On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said: snip So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? I am wondering if anyone has experience with the Audiocodes MP114 (2fxs/2fxo)? This is pricey, but I am SO sick of mucking around with consumer grade bs (ie grandstream). I am also curious about the mediatrix 1204 and the Multitech MVP-130 although it sort of looks to me like the multitech doesn't do SIP. Any thoughts or help, before I take an expensive leap? Marty PS asterisk 1.2.13 ___ --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
[asterisk-users] Re: Loosing IAX connection between offices
On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. For trunking, avoid IAX and use SIP. ___ --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
[asterisk-users] Digium TE407P vs. Sangoma A104d
Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is 5 stars! Awesome! It Rocks! They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I'm interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --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
RE: [asterisk-users] asterisk manager originate command
I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Are you wanting the uniqueid for the call right after it is started, i.e., while it is still in progress? What is in your Dial command? -MC ___ --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
Re: [asterisk-users] Re: Loosing IAX connection between offices
Louis-David Mitterrand wrote: On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote: Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. For trunking, avoid IAX and use SIP. For the record I have had 4 asterisk (1.2.6) boxes connected with IAX trunks (every box registers with every other box) for over 6 months now with no issues. I did have some issues with a Linksys firewall, but that was with SIP registrations, not IAX. A firmware update fixed that. I'm no * dev, but what exactly is so crap about IAX? -Chris ___ --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
RE: [asterisk-users] Digium TE407P vs. Sangoma A104d
I'd like to hear the feedback too... specific comparison interests: * echo cancellation effectiveness (without introducing artifacts) * ease of provisioning * quality of wiki other docs for giving detailed guidance/hints * customer support from vendor when you encounter problems * PRI protocol variant and IE support (e.g. name display) * Q.SIG support for path replacement * CAS protocol support (EM winkstart? Feature-group D to get ANI?) * load placed on host * resilience to overheating (e.g. in poorly cooled wiring closets) * Heat/BTU output Thanks, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Monday, December 04, 2006 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digium TE407P vs. Sangoma A104d Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is 5 stars! Awesome! It Rocks! They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I'm interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --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
Re: [asterisk-users] asterisk manager originate command
There is no dial command, I'm sending originate action from asterisk manager. Michael Collins wrote: I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Are you wanting the uniqueid for the call right after it is started, i.e., while it is still in progress? What is in your Dial command? -MC ___ --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
Re: [asterisk-users] Digium TE407P vs. Sangoma A104d
We have two of the TE407P's active in one of our gateways, and it's been an awesome card to deal with. I'd definitely have to say, once we installed those, the server's been extremely stable, the T1's have had no echo problems, it just works. I'd recommend the Digium cards. Disclaimer: I've never used the Sangoma cards. On Mon, 2006-12-04 at 12:21 -0800, Michael Collins wrote: Has anyone had experience with one or both of these cards? I’m in a position where I might need to recommend one over the other. I’ve read everything that I can find online, so now I’d like to hear of personal experiences. Everything I read on both cards is “5 stars! Awesome! It Rocks!” They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I’m interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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
[asterisk-users] Re: Odd queue issue
Debug of the sip peer 126 shows: -- Called 126 -- Agent/ is ringing Retransmitting #1 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - What is doing this? ___ --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
[asterisk-users] Re: RE : Re: Recommendation for FXO
On 2006-12-04 11:54:14 -0800, [EMAIL PROTECTED] said: Hi the list and Marty, Take a look to www.aliwei.com. Thanks for the idea, but this looks CHILLINGLY identical to the wellgate? I wonder if this is the same hardware with a different name on it? Is this something you are using personally (in particular the FXO and and asterisk)? Thanks again, Marty Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 4 décembre 2006 20:47 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: Recommendation for FXO On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said: snip So, I would like to purchase another PSTN gateway which WORKS WELL with asterisk. I need it to hook up via ethernet, since my platform of choice (mac OSX) has no PCI card support. I only have one PSTN line, and already have other ATA's for FXSs, so I really only need one FXO port, although I realize there is no such animal. Any positive experiences with FXO gateways that connect via ethernet? Especially with a long loop/echo issues (ie not SPA3000)? I am wondering if anyone has experience with the Audiocodes MP114 (2fxs/2fxo)? This is pricey, but I am SO sick of mucking around with consumer grade bs (ie grandstream). I am also curious about the mediatrix 1204 and the Multitech MVP-130 although it sort of looks to me like the multitech doesn't do SIP. Any thoughts or help, before I take an expensive leap? Marty PS asterisk 1.2.13 ___ --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 ___ --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
[asterisk-users] Re: Nokia E60 problems
On 2006-12-04 04:10:36 -0800, Giedrius Augys [EMAIL PROTECTED] said: Hi, I am testing Nokia E60 with Asterisk. And I noticed that if another side is busy, nokia is still calling (I hear alerting), it do not show that another side is busy. Maybe somebody has noticed the same problem too adnd solved this one. I made the same tests with Xlite and don't have any problems like nokia. Please help me Not here, my e60 says: Extension busy or something similar. Try looking over your config on the asterisk side again? dunno. Marty ___ --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
RE: [asterisk-users] asterisk manager originate command
There is no dial command, I'm sending originate action from asterisk manager. Oops, I didn't ask my question correctly. You're right, it isn't a dial command. What I wanted to know was the contents of your originate action, e.g.: Channel= 'zap/g0/' . $dialed_num (From one of my Perl scripts using POE::Component::Client::Asterisk::Manager) Thanks, MC ___ --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
Re: [asterisk-users] asterisk manager originate command
My code is using phpagi-asmanagerbut what is sent is... Action: Originate Channel: SIP/802 Context: from-internal Exten: number to dial Priority: 1 Callerid: 802 Michael Collins wrote: There is no dial command, I'm sending originate action from asterisk manager. Oops, I didn't ask my question correctly. You're right, it isn't a dial command. What I wanted to know was the contents of your originate action, e.g.: Channel= 'zap/g0/' . $dialed_num (From one of my Perl scripts using POE::Component::Client::Asterisk::Manager) Thanks, MC ___ --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
Re: [asterisk-users] Help with IAX Trunk
On 11:02, Mon 04 Dec 06, Dave Morrow wrote: Yes. That was the solution. Not sure why that 'r' is there in the first place It's there to provide 'ringing' indication on links that do not provide it. (voip-voip connections or voip-pri|pstn connections without early-media passthru) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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
[asterisk-users] Re: 200+ analog phones connected to FXS modules
CG == Csibra Gergo [EMAIL PROTECTED] writes: CG Well, I think there's far more htan 1000 interrupts come from an CG T1/E1 card. Or do you think 1000/channel? No I mean 1000 interrupts total, across all channels on all cards. Otherwise the driver is just broken. Of course in a sane system there would be no card interrupts at all, just polling. Linux wasn't sane enough for that until relatively recently, so we're stuck with interrupts. Why would you not handle all channels in just one interrupt? /Benny ___ --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
Re: [asterisk-users] How can i processed with Call Snooping,
How can i Processed the call Snooping, it my fifth Requesting and posting to Users, Nobody replies it,,, see http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy hth ___ --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
[asterisk-users] Connecting Asterisk to an NEC Aspire
All New Asterisk user, wondering if anybody has connected an Asterisk box to an NEC Aspire S? We're in the beginning processes of attempting this, we'd like to have the Asterisk box connected as an extension off of the NEC box, wondering about the wiring and settings/programming needed to get the units talking to each other. Any help/info appreciated Mike ___ --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
RE: [asterisk-users] Connecting Asterisk to an NEC Aspire
New Asterisk user, wondering if anybody has connected an Asterisk box to an NEC Aspire S? We're in the beginning processes of attempting this, we'd like to have the Asterisk box connected as an extension off of the NEC box, wondering about the wiring and settings/programming needed to get the units talking to each other. Analog or digital stations? -MC ___ --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
[asterisk-users] T1 PRI not announce this is long distance call, please add 1 for this call...
Can any one help? In Toronto, we can't identify if a number is long distance call or not. If long distance call, we have to prefix with 1. We should hear a voice prompt as above to indicate that it is not a local call. However, we hear the normal ring back tone (indicating the phone had been connected, but actually not) when we call this long distance call without prefixing 1. Here is the message shown in CLI. Requested transfer capability: 0x00 - SPEECH -- Called g0/9056671191 -- Zap/1-1 is proceeding passing it to SIP/9188-0e6a -- PROGRESS with cause code 127 received -- Zap/1-1 is making progress passing it to SIP/9188-0e6a Thanks in advances. Isaac ___ --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
RE: [asterisk-users] Connecting Asterisk to an NEC Aspire
On the NEC, digital stations (ip1na-12txh) md -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Monday, December 04, 2006 3:38 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Connecting Asterisk to an NEC Aspire New Asterisk user, wondering if anybody has connected an Asterisk box to an NEC Aspire S? We're in the beginning processes of attempting this, we'd like to have the Asterisk box connected as an extension off of the NEC box, wondering about the wiring and settings/programming needed to get the units talking to each other. Analog or digital stations? -MC ___ --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
Re: [asterisk-users] ASterisk and SER
What is the purpose of that sort of call routing, it does seem like a loop to me. Asterisk is probably getting re-invited to itself... On 12/4/06, Arun Kumar [EMAIL PROTECTED] wrote: HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun ___ --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
Re: [asterisk-users] asterisk manager originate command
currently thats not possible unless you speciffy the async flag, in that case Event: OriginateSuccess or Event: OriginateFailed event will be launched with the uniqueid Regards On 12/4/06, Rodrigo Gonzalez [EMAIL PROTECTED] wrote: My code is using phpagi-asmanagerbut what is sent is... Action: Originate Channel: SIP/802 Context: from-internal Exten: number to dial Priority: 1 Callerid: 802 Michael Collins wrote: There is no dial command, I'm sending originate action from asterisk manager. Oops, I didn't ask my question correctly. You're right, it isn't a dial command. What I wanted to know was the contents of your originate action, e.g.: Channel= 'zap/g0/' . $dialed_num (From one of my Perl scripts using POE::Component::Client::Asterisk::Manager) Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I am reading about xPL protocol since [EMAIL PROTECTED] 0.9, when I first used it. Its been more than two years now and I never saw any documentation on it. Their website itself needs material to be put on it. So xML is not a useful thing at all at this point. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I meant xPL, not xML in my last eamil. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 PRI not announce this is long distance call, please add 1 for this call...
do something like this in your extensions.conf:
exten = _NXXNXX,1,Dial(ZAP/g0/1{$EXTEN})
exten = _222NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _223NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _224NXX,1,Dial(ZAP/g0/{$EXTEN})
Where 222, 223 and 224 are local area codes.
On 12/4/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Can any one help? In Toronto, we can't identify if a number is long
distance call or not. If long distance call, we have to prefix with 1. We
should hear a voice prompt as above to indicate that it is not a local call.
However, we hear the normal ring back tone (indicating the phone had been
connected, but actually not) when we call this long distance call without
prefixing 1.
Here is the message shown in CLI.
Requested transfer capability: 0x00 - SPEECH
-- Called g0/9056671191
-- Zap/1-1 is proceeding passing it to SIP/9188-0e6a
-- PROGRESS with cause code 127 received
-- Zap/1-1 is making progress passing it to SIP/9188-0e6a
Thanks in advances.
Isaac
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Re: [asterisk-users] Is there any Asterisk controllable thermostat?
Quoting Zeeshan Zakaria [EMAIL PROTECTED]: I am reading about xPL protocol since [EMAIL PROTECTED] 0.9, when I first used it. Its been more than two years now and I never saw any documentation on it. Their website itself needs material to be put on it. So xML is not a useful thing at all at this point. I have done some work on snmp enabling thermostats and burglar alarm systems with the end goal to integrate with asterisk / web interface eventually. SNMP in general is well supported and well documented for controlling devices. A subagent for the particular hardware and a generic snmp interface for asterisk would be what is required. The devices I was working with are actually talked to physically by the Dallas 1-wire hardware/protocol, and there is various linux support already for talking to those type of busses. interfacing to asterisk could be a nice module that talks snmp, or it could be as simple as calling the existing shell commands from netsnmp such as snmpget and snmpset. If anyone wants to discuss with me offlist, I have official IANA enterprise numbers already allocated for this, as well as some of the mibs started which cover a lot more than just the thermostats. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
Quoting Zeeshan Zakaria [EMAIL PROTECTED]: I meant xPL, not xML in my last eamil. yeah got that :) - and I agree with you, it seems more like vapourware than anything with much substance at this point, I just looked over that a few days ago when I ran across it by accident. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Answer a call that is not ringing on yourextension
Another solution is to use the Pickup() command. It will pick up a call on a specific extension that is in the ringing state: [Description] Pickup([EMAIL PROTECTED]): This application can pickup any ringing channel that is calling the specified extension. If no context is specified, the current context will be used. For example, my co-workers extension is 203. I hear his phone ringing, and I dial my pre-defined pickup extension (**203) to pickup his call. Dialplan example: Exten = **203,1,Pickup(203) Exten = **203,2,Hangup() Note: the read I use ** is GXP-2000 phones will dial **exten while the BLF light is in ringing state. You can use whatever you want. Wes Baehr -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ove Aursand Sent: Monday, December 04, 2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Answer a call that is not ringing on yourextension David Parcerisa wrote: Answer a call that is not ringing on your extension. I want to pick up an external call that is ringing on another extension that is not mine. Now in my old standard pbx I press 5 and I get the call. How to do this with asterisk? thank you. ___ --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 Hi David You can use callgroup/pickupgroup in sip.conf, and use features.conf to choose what you have to press on the phones to pickup other calls. All phones have to be in the same group (0-64 I think). Regards, Ove ___ --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
Re: [asterisk-users] Re: Odd queue issue
Could you explain which devices have what IP and what is behind NAT between what? On 12/4/06, Matt [EMAIL PROTECTED] wrote: Debug of the sip peer 126 shows: -- Called 126 -- Agent/ is ringing Retransmitting #1 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (NAT) to 63.174.244.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 63.174.244.175:5060;branch=z9hG4bK598573e4;rport From: Test VoIP Accounts sip:[EMAIL PROTECTED];tag=as1a3a38f5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 04 Dec 2006 20:42:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 275 v=0 o=root 3555 3555 IN IP4 63.174.244.175 s=session c=IN IP4 63.174.244.175 t=0 0 m=audio 19720 RTP/AVP 0 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - What is doing this? ___ --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
Re: [asterisk-users] T1 PRI not announce this is long distance call, please add 1 for this call...
Using the PSTN in Toronto ie 416 NXX X all calls to 647 and 416
exchanges are local. 905 is an over-lapping area code, most excahnges are
local, however Whitby (905 430 ) is Long Distance while 416 428
(Ajax) is not. You can find out which ones are long distance (from the
CRTC web site) and modify your dial plan to add the 1 to the dialed number
or route the numbers to a DID with your friendly ITSP like Unlimitel for
termination.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
do something like this in your extensions.conf:
exten = _NXXNXX,1,Dial(ZAP/g0/1{$EXTEN})
exten = _222NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _223NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _224NXX,1,Dial(ZAP/g0/{$EXTEN})
Where 222, 223 and 224 are local area codes.
On 12/4/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Can any one help? In Toronto, we can't identify if a number is long
distance call or not. If long distance call, we have to prefix with 1.
We
should hear a voice prompt as above to indicate that it is not a local
call.
However, we hear the normal ring back tone (indicating the phone had
been
connected, but actually not) when we call this long distance call
without
prefixing 1.
Here is the message shown in CLI.
Requested transfer capability: 0x00 - SPEECH
-- Called g0/9056671191
-- Zap/1-1 is proceeding passing it to SIP/9188-0e6a
-- PROGRESS with cause code 127 received
-- Zap/1-1 is making progress passing it to SIP/9188-0e6a
Thanks in advances.
Isaac
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Re: [asterisk-users] Is there any Asterisk controllable thermostat?
I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug ___ --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
[asterisk-users] any possibility of Vonage Integration
Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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
RE: [asterisk-users] any possibility of Vonage Integration
To be more elaborate, i am using 10 vonage lines in my office, can i connect them all using asterisk, or is it possible to configure those accounts on asterisk instead of the linksys boxes i am using. Regards Vijay Gandhi -Original Message- From: Vijay Gandhi [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] any possibility of Vonage Integration Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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
RE: [asterisk-users] Connecting Asterisk to an NEC Aspire
On the NEC, digital stations (ip1na-12txh) I am not familiar with the Aspire, but if it is even remotely like the 2400 then you might be able to get a jumpstart using my 2400 how-to: http://www.voip-info.org/wiki/index.php?page=Asterisk+NEAX2400 It deals with getting a Tormenta2 clone talking to a 2400 station side T1 card. If you know an NEC tech familiar with both PBX's then he might be able to translate this into something you can use. Hope this helps and good luck! -MC ___ --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
RE: [asterisk-users] How to park calls on a specific extension
On Fri, 1 Dec 2006, Ken Williams wrote: I was able to set a program to speed dial the park extension. Then a user just hits TNFR followed by the line I've programmed to speed dial park. Heh. That's what I want to do, but when I dial the ValetParking extension now, not only do I not park the call, my ssh session to the Asterisk box freezes. I need to make sure I have a version of app_valetparking.so that is compatible with my Asterisk/Trixbox build. If you get the HOLD button to do this, I'd love to hear how :). From: Steve Sobol Sent: Fri 12/1/2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to park calls on a specific extension On Thu, 30 Nov 2006, Brad Templeton wrote: Problem there is only some phones have line buttons, and when they have them they are scarce and there's many things you might like to do with them, and dedicating them to this would be low on my list. Dedicating one speed Eventually I am going to do a little sleuthing to find out what my GXP-2000s' HOLD buttons send to Asterisk, and I'm going to make the HOLD button park a call. :) Until then, I'm going to have to use an interim solution. Isn't there a separate hold dialplan context? -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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
[asterisk-users] How to stop Asterisk to pick up incoming PSTN signal
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the functionality to make the call out? Thanks Gidean___ --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
[asterisk-users] RESEND: Blind transfer # not working for forwarded or picked calls
Resending this since I got no response Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 Kind regards Roger ___ --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
Re: [asterisk-users] any possibility of Vonage Integration
You can add Vonage accounts to your asterisk. The only account that Vonage will let you use is there Biz account higher rates. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Vijay Gandhi wrote: Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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 Inbound (clean). Database: 0653-2, 12/04/2006 - 12/5/2006 12:58:38 AM ___ --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
