Re: [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!
On Sat, 2008-01-12 at 08:35 -0600, [EMAIL PROTECTED] wrote: Date: Sat, 12 Jan 2008 11:02:17 +0100 From: Johansson Olle E [EMAIL PROTECTED] Subject: [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration! To: Asterisk List - Non-Commercial Discussion Users Mailing asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes I've written a new article about Asterisk 1.4's Jabber integration. Check it out at http://www.voip-forum.com/asterisk/2008-01/xmpp/ /Olle [from http://www.voip-forum.com/asterisk/2008-01/xmpp/ ]: * Jabber presence support in the dialplan: By letting your Asterisk connect to a Jabber server by using a Jabber account, you can add buddies to that account and check the buddies presence in the Asterisk dialplan. This way, call routing decisions can be based on the status of Jabber accounts. (...) * Asterisk as a Jabber module: In a more advanced mode, Asterisk can register itself as a module to your Jabber server (as a Jabber component). This mode means better integration to Jabber, but requires more from the Jabber clients. [/from] Can an Asterisk server hold logins for multiple Japper accounts on a remote Jabber server, and carry multiple Jabber calls simultaneously the way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is each of those Jabber calls as lightweight as, say, each SIP call? If not, is there a way to increase the capacity of Asterisk to carry about as many Jabber calls as it can carry SIP calls? -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MRCP Asterisk Integration
I think such feature should rather come from VXML server but I may be wrong. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interconnecting an asterisk server with an old alcatel PBX through a Digium B410P
www.alcatelunleashed.com is full of info for that kind of topic. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adtran 750 and EM Wink
Hi List, I have some T1 750s; hookup via a T1 to a TE420P card. We have them all set as EM wink trunks. We get a dial tone and can call numbers, but when i call another number the ringing I hear on my phone is very weird, its not the standard ring tone. What will cause this? Is this a * setting or something in the Adtran I am fixing, I just want good ole ringing to be heard! Any help would be great! Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC
clive.chan(Atn) wrote: Hi all, Just want to check from the list experienced personal about the Digium HPEC, where I had purchased the HPEC and wish to run with TDM card Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I had changed my hpec file from i686 to i386. The error that I had as bellow; Found key 'HPEC-XX' for 2 channels. Found valid HPEC licenses for 2 channels. Failed to get license challenge: No such device Any advice well be welcome. I must confess, I've been meaning to give HPEC a try (especially since I apparently can get a free license having purchased a fully loaded TDM400), but I've not got round to it. I've had OSLEC running on my home test machine, does anyone have any idea how the performance compares? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about queues and the definition and agents
Paul wrote
;Pause/unpause Queue
exten = 424,1,PauseQueueMember(|SIP/${CALLERID(num)})
exten = 424,2,Playback(unavailable)
exten = 424,3,Hangup
exten = 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)})
exten = 425,2,Playback(available)
exten = 425,3,Hangup
Following your suggestion and a number of postings and articles I have
configured the following:
exten = 6662,1,ADDQUEUEMEMBER(queue1|)
;exten = 6662,1,ADDQUEUEMEMBER(queue1|SIP/${ID${CALLERID(num)[EMAIL PROTECTED])
exten = 6662,2,HANGUP()
When I now call 6662 and then enter show queues in the cli, I get:
queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 60s
Members:
SIP/sguenther (dynamic) (Unavailable) has taken no calls yet
No Callers
Why is the agent unavailable and which context would asterisk use to
call the agent?
Here's the output of the login:
-- Executing [EMAIL PROTECTED]:1] AddQueueMember(SIP/sguenther-08202ac8,
queue1|) in new stack
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/sguenther-08202ac8, ) in new
stack
== Spawn extension (local, 6662, 2) exited non-zero on
'SIP/sguenther-08202ac8'
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
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[asterisk-users] problems with zaptel and Udev
Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl-AGI process
Abdul wrote:
routes.pl
$dgw = 'SIP/5556'; #A-Z carrier
$opt = 'L(6:1)';
$AGI-set_variable(routecall-destination, $dgw);
$AGI-set_variable(routecall-args, $opt);
Extnenitons.conf
[testwell]
exten = _x.,1,Set(TIMEOUT(absolute)=3660)
exten = _x.,2,AGI(routes.pl)
exten = _x.,3,Dial(${routecall-destination},${routecall-args})
exten = h,1,DeadAGI(stop.pl)
Warnning :
[Jan 12 14:34:22] WARNING[27323]: app_dial.c:863 dial_exec_full: Dial
requires an argument (technology/number)
== Spawn extension (testwell, 9745424620, 9) exited non-zero on
'SIP/8098179675-b726f5e8'
Hmm. You might be having problems with escaping. That's one thing I
recall bothered me about Asterisk::AGI
The best way to debug it is to add NoOp(routecall-destination =
${routecall-destination}) to your dial plan and look at the verbose
output to see what it's saying. Then try adding some escapes to your
Perl script and see if that helps.
That's first a single quote ' then a double quote as below, and to
close it obviously a double then a single:
$dgw = 'SIP/5556'; #A-Z carrier
$opt = 'L(6:1)';
--
Real CNAM data for incoming Caller ID @ www.cnam.info
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Re: [asterisk-users] problems with zaptel and Udev
On Sun, Jan 13, 2008 at 05:33:58PM +, robert boardman wrote: Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Hmm is it udev that modprobes the modules on the PCI bus? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports and CentOS firewall
Check this out: http://www.voip-info.org/wiki-Asterisk+firewall+rules dave cantera wrote: ed, this may be somewhat liberal but should do the trick... daveC -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 69 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 69 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5061 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5062 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m tcp --dport 5038 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5036 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 5004 -j ACCEPT # -A RH-Firewall-1-INPUT -p icmp -m icmp --icmp-type any -j ACCEPT -A RH-Firewall-1-INPUT -p ipv6-crypt -j REJECT -A RH-Firewall-1-INPUT -p ipv6-auth -j REJECT -A RH-Firewall-1-INPUT -d 224.0.0.251 -p udp -m udp --dport 5353 -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m state --state NEW -m tcp --dport 80 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited Ed Nunez wrote: If I enable the firewall on my Server, which ports should I open for Asterisk to work properly. Is it enough to just open the SIP ports? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with zaptel and Udev
Tzafrir Cohen wrote: On Sun, Jan 13, 2008 at 05:33:58PM +, robert boardman wrote: Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Hmm is it udev that modprobes the modules on the PCI bus? yes I think it is , I'll re complie zaptel to see if that makes any difference Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ProxyPal for AMI Proxy Development
Hi all, I'm writing a real-time (not RealTime) proxy server for the AMI interface. Although I'll be using it for some commercial products, the proxy software itself will be released under GPL. I was wondering if there would be any interest in testing it from the community? I don't have access to a high (or even medium volume) system as this office only has 4 extensions with maybe 50 calls a day. My particular needs for this software: * Real-time, event driven interface to Asterisk AMI, no polling. (done.) * Event Filtering. (somewhat done.) * Various Decorator plugins to customize packets/requests. (plain and xml done.) * Easy configuration of users that mimic existing Asterisk AMI permissions model. (almost done.) Any thoughts, suggestions or comments welcome. -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ProxyPal for AMI Proxy Development
astmanproxy does this already, I think .. Julian. Lee Jenkins wrote: Hi all, I'm writing a real-time (not RealTime) proxy server for the AMI interface. Although I'll be using it for some commercial products, the proxy software itself will be released under GPL. I was wondering if there would be any interest in testing it from the community? I don't have access to a high (or even medium volume) system as this office only has 4 extensions with maybe 50 calls a day. My particular needs for this software: * Real-time, event driven interface to Asterisk AMI, no polling. (done.) * Event Filtering. (somewhat done.) * Various Decorator plugins to customize packets/requests. (plain and xml done.) * Easy configuration of users that mimic existing Asterisk AMI permissions model. (almost done.) Any thoughts, suggestions or comments welcome. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with zaptel and Udev
I had the same issue and updated my Zaptel drivers to version 1.4.17 and it's rebooting fine now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of robert boardman Sent: Sunday, January 13, 2008 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] problems with zaptel and Udev Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ProxyPal for AMI Proxy Development
Julian Lyndon-Smith wrote: astmanproxy does this already, I think .. Julian. Of course ;) AstManProxy is a great product from what I had read up on it. One thing is that it requires (if I'm not mistaken) an mysql installation which is too heavy of a dependency for some applications that I have in mind to write. The proxy I'm writing allows real-time traffic between proxy clients and the Asterisk AMI and the traffic is cached maybe 50-100ms before being sent to either end so there is no need to save the packet traffic to a database. For me, I need something lean in terms of 3rd party software and lean on memory with a simple deployment of an executable and maybe a few config files. I figured since I was writing it anyway, I'd just release it to the community... I'll post when its ready for testing or usage along with link to sources. Thanks again, -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about queues and the definition and agents
Maybe ADDQUEUEMEMBER(queue1|${CALLERID(num)) is closer to what you are
looking foror ADDQUEUEMEMBER(queue1) - without the pipe...
PaulH
On Sun, 2008-01-13 at 18:17 +0100, Stefan Guenther wrote:
Paul wrote
;Pause/unpause Queue
exten = 424,1,PauseQueueMember(|SIP/${CALLERID(num)})
exten = 424,2,Playback(unavailable)
exten = 424,3,Hangup
exten = 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)})
exten = 425,2,Playback(available)
exten = 425,3,Hangup
Following your suggestion and a number of postings and articles I have
configured the following:
exten = 6662,1,ADDQUEUEMEMBER(queue1|)
;exten = 6662,1,ADDQUEUEMEMBER(queue1|SIP/${ID${CALLERID(num)[EMAIL
PROTECTED])
exten = 6662,2,HANGUP()
When I now call 6662 and then enter show queues in the cli, I get:
queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 60s
Members:
SIP/sguenther (dynamic) (Unavailable) has taken no calls yet
No Callers
Why is the agent unavailable and which context would asterisk use to
call the agent?
Here's the output of the login:
-- Executing [EMAIL PROTECTED]:1] AddQueueMember(SIP/sguenther-08202ac8,
queue1|) in new stack
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/sguenther-08202ac8, ) in new
stack
== Spawn extension (local, 6662, 2) exited non-zero on
'SIP/sguenther-08202ac8'
Thanks for your help,
Stefan
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[asterisk-users] Aastra Venture
Does anyone on-list have any experience with this system? I'd like to speak with someone who's tried it out. Thanks, Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call parking
Is there a good way to set the callerid(name) for calls being returned from parking? We tried using the parkandannounce function, but we couldn't get the audio to play back nicely. (we don't want the park position returned as a separate phone call...) ideas? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ProxyPal for AMI Proxy Development
Lee Jenkins wrote: Julian Lyndon-Smith wrote: astmanproxy does this already, I think .. Julian. Of course ;) AstManProxy is a great product from what I had read up on it. One thing is that it requires (if I'm not mistaken) an mysql installation which is too heavy of a dependency for some applications that I have in mind to write. It does not require an mysql of any type at all. The proxy I'm writing allows real-time traffic between proxy clients and the Asterisk AMI and the traffic is cached maybe 50-100ms before being sent to either end so there is no need to save the packet traffic to a database. For me, I need something lean in terms of 3rd party software and lean on memory with a simple deployment of an executable and maybe a few config files. astmanproxy executable and astman.conf are the only files I figured since I was writing it anyway, I'd just release it to the community... I'll post when its ready for testing or usage along with link to sources. Cool. choice is always a fine thing. Thanks again, Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
