Re: [asterisk-users] Queue not sending call to Agent
Provide the entry for Agent SIP/9013XX9XX8 along with parameters
'callcounter' and 'qualify' from sip.conf.
Also provide CLI outputs of 'core show channels',sip show peers' and 'queue
show' when...
(1)First caller enters the Queue
(2)First caller gets connected with Agent
(3)First caller gets disconnected from Agent
(4)Second caller enters the Queue
You may have sequences changed for step no 3 and 4 in your scenario.
[SATISH]
On Sat, Jun 11, 2011 at 2:56 AM, wrote:
> Queue not sending call to Agent
>
>
>
> I am having an issue and i am not sure if it is a bug or a config issue. I
> was originally running Asterisk 1.8.1.1 when I noticed this issue. I
> upgraded to 1.8.4.2 to see if that would fix it but it didn't.
>
> The issue is that I have a call queue and the agent dials a number to log
> into the queue. When someone calls the queue the first time the call is sent
> to the agent without issue. The issue is that any calls after the first are
> placed in the queue and never sent to the agent who is logged in and
> available. Before I call the queue I do a "show queue" and it shows the
> agent as
>
> Asterisk18*CLI> queue show
> irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime,
> 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet
> No Callers
>
>
> Then the call comes into the queue and the callee just sits in the queue.
> When I do a "show queue" again when the callee is in the queue it shows the
> agent as busy
> Asterisk18*CLI> queue show
> irock.com has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime,
> 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> SIP/9013XX9XX8 (dynamic) (Busy) has taken no calls yet
> Callers:
> 1. SIP/9013XX9XX8-0001 (wait: 0:12, prio: 0)
>
>
> So I am not sure what happened because the agent was free before the call.
> If I do a reload at the Asterisk CLI and then call again the agent gets the
> call and then the second call is once again placed in the queue. I will
> attach a SIP Debug that shows what is going on. I don't see any SIP invites
> leaving Asterisk to invite the agent to the call.
>
> One other thing Currently in my config I have the agent show up as just
> the username which is the phone number. If I set it so that the agent shows
> up as phonenumber@blah then I can call the agent constantly without any
> issue. The only problem here is that when I do a "queue show" the agent
> shows up as "unknown" status. So when the agent is on a call and someone
> else calls the agent will be interrupted.
>
>
>
> This is what I have in queues.conf
> [irock.com]
> strategy=ringall
> ringinuse=no
> joinempty=yes
> leavewhenempty=no
> announce-frequency=30
> min-announce-frequency=15
> periodic-announce-frequency=60
> announce-holdtime=yes
> announce-position=yes
>
> ; ("You are now first in line.")
> queue-youarenext = queue-youarenext
> ; ("There are")
> queue-thereare = queue-thereare
> ; ("calls waiting.")
> queue-callswaiting = queue-callswaiting
> ; ("The current est. holdtime is")
> queue-holdtime = queue-holdtime
> ; ("minutes.")
> queue-minutes = queue-minutes
> ; ("seconds.")
> queue-seconds = queue-seconds
> ; ("Thank you for your patience.")
> queue-thankyou = queue-thankyou
> ; ("Hold time")
> queue-reporthold = queue-reporthold
> ; ("All reps busy / wait for next")
> periodic-announce = queue-periodic-announce
>
>
>
> This is what I have in extensions.conf
> exten => 9012XX1XX1,1,Answer()
> exten => 9012XX1XX1,n,Set(QUEUE_MAX_PENALTY=0);
> exten => 9012XX1XX1,n,Queue(irock.com,t)
> exten => 9012XX1XX1,n,Hangup()
>
> exten => *50,1,Answer
> exten => *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
> exten => *50,n,Hangup
>
> exten => *51,1,Answer
> exten => *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
> exten => *51,n,Hangup
>
> [macro-queue-login]
> exten => s,1,Set(agent=${EXTEN:4})
> exten => s,n,Set(queue=irock.com)
> exten => s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
> exten => s,n,AddQueueMember(${queue});
> exten => s,n,Playback(agent-loginok)
>
> [macro-queue-logout]
> exten => s,1,Set(agent=${EXTEN:4})
> exten => s,n,Set(queue=irock.com)
> exten => s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
> exten => s,n,RemoveQueueMember(${queue});
> exten => s,n,Playback(agent-loggedoff)
> --
> _
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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h
Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files
ROFL On Sun, Jun 12, 2011 at 6:27 PM, Alex Balashov wrote: > And that means he can be replaced with a small shell script. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jun 12, 2011, at 5:59 PM, C F wrote: > >> You gotto love this this guy. You can almost predict what his next question >> is. >> >> On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad wrote: >>> Hi All; >>> >>> I need to create the needed files for the Cisco Phones to be placed in the >>> TFTP server to be able to register on Asterisk. >>> >>> I need a help in the following please: >>> >>> 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP >>> address of Asterisk? >>> >>> 2) Regarding to the file: SIP.cnf, if I need to let it appear >>> at the Phone (the first line for example) the extension, so this will be >>> the line1_name? For example, if I need the extension to be 700 then I set >>> line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : >>> 701? >>> >>> 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP >>> address of Asterisk, but where? What is the format to write the Asterisk IP >>> address in this file? >>> >>> 4) About the dialplan.xml file, what the below means? >>> >>> >>> >>> Any help? >>> Regards >>> Bilal >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> 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 -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tls/srtp: sip_xmit error: returned -2
I'm still no further advanced on this, but I think I have narrowed it down to tls. I have sip debug logs which shows that the server cannot contact the tls enabled phone at the same time this error crops up. The log says "calling " and then the error. With TLS disabled, though, SRTP still doesn't work either though. I have no knowledge of how to move forward on this, so some pointers would be very much appreciated. On 06/07/11 12:11, Da Rock wrote: I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up: [Jun 7 11:44:13] NOTICE[88483]: chan_sip.c:19842 handle_response_peerpoke: Peer '' is now Reachable. (171ms / 2000ms) [Jun 7 11:46:17] NOTICE[88483]: chan_sip.c:25072 sip_poke_noanswer: Peer '' is now UNREACHABLE! Last qualify: 203 [Jun 7 11:46:29] NOTICE[88483]: chan_sip.c:19842 handle_response_peerpoke: Peer '' is now Reachable. (1888ms / 2000ms) When I call on this phone I get: [Jun 7 11:40:47] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2: Invalid argument [Jun 7 11:41:01] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2: Invalid argument [Jun 7 11:41:15] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2: Invalid argument [Jun 7 11:41:29] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:36129 returned -2: Invalid argument -- Registered SIP '' at 192.168.0.200:57805 [Jun 7 11:41:31] NOTICE[88483]: chan_sip.c:19842 handle_response_peerpoke: Peer '' is now Reachable. (10ms / 2000ms) When I call from another phone I get: [Jun 7 11:55:30] NOTICE[88483]: chan_sip.c:25072 sip_poke_noanswer: Peer '' is now UNREACHABLE! Last qualify: 13 -- SIP/-0024 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/-0023' status is 'CONGESTION' [Jun 7 11:56:22] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2c992000 (len 599) to 192.168.0.200:45931 returned -2: Interrupted system call and eventually: [Jun 7 11:57:46] WARNING[88483]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x2cefb000 (len 599) to 192.168.0.200:45931 returned -2: Unknown error: 0 I'm using my own CA setup for purposes beyond just this need, so I'm using openssl commands directly and everything works elsewhere- so my CA setup is fine (includes SAN). My config for tls/srtp looks like this (remember, the rest works very happily): [global] encryption = yes tlsenable = yes tlsbindaddr = 0.0.0.0 tlscertfile = /path/to/asterisk/certificate/and/key/in/a/single/file tlscafile = /path/to/CA/certificate tlscipher = ALL tlsclientmethod = tlsv1 [tls user] transport=tls Can someone give me any clues to what is happening? I've checked my packet flow with tcpdump and wireshark as well, but I'm still left mystified. Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco
On Sun, Jun 12, 2011 at 3:18 PM, Steve Edwards
wrote:
> On Sun, 12 Jun 2011, bilal ghayyad wrote:
>
>> How can I see the the files are now loaded? Normally in the windows, I can
>> see that a request reached to the TFTP server and the files now are loaded
>> (upload or download)?
>
> The tftp daemon should log the transfers via syslogd. For example:
>
> Jun 12 12:03:23 dt in.tftpd[11486]: RRQ from 192.168.0.19 filename
> /spa841.cfg
>
> I start tftp via xinetd and the configuration looks like this:
>
> service tftp
> {
> cps = 100 2
> bind = 192.168.0.1
> disable = no
> flags = IPv4
> per_source = 11
> protocol = udp
> server = /usr/sbin/in.tftpd
> server_args = -c -s /tftpboot -v -v -v -v -v -v
> socket_type = dgram
> user = root
> wait = yes
> }
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
You should really be asking on the Fedora lists or even Google, this
is about TFTP on Fedora, not anything to do with Asterisk.
http://tinyurl.com/6xj3he3
http://tinyurl.com/3hjtwsu
Also, check /var/log/messages on your server. Are there any tftp messages there?
Also run tftpd with the "-v" flag (multiple times -- try "man tftpd").
Thanks,
Steve Totaro
--
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Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files
And that means he can be replaced with a small shell script. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 12, 2011, at 5:59 PM, C F wrote: > You gotto love this this guy. You can almost predict what his next question > is. > > On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad wrote: >> Hi All; >> >> I need to create the needed files for the Cisco Phones to be placed in the >> TFTP server to be able to register on Asterisk. >> >> I need a help in the following please: >> >> 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP >> address of Asterisk? >> >> 2) Regarding to the file: SIP.cnf, if I need to let it appear at >> the Phone (the first line for example) the extension, so this will be the >> line1_name? For example, if I need the extension to be 700 then I set >> line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : >> 701? >> >> 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP >> address of Asterisk, but where? What is the format to write the Asterisk IP >> address in this file? >> >> 4) About the dialplan.xml file, what the below means? >> >> >> >> Any help? >> Regards >> Bilal >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files
You gotto love this this guy. You can almost predict what his next question is. On Sun, Jun 12, 2011 at 5:16 PM, bilal ghayyad wrote: > Hi All; > > I need to create the needed files for the Cisco Phones to be placed in the > TFTP server to be able to register on Asterisk. > > I need a help in the following please: > > 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP > address of Asterisk? > > 2) Regarding to the file: SIP.cnf, if I need to let it appear at > the Phone (the first line for example) the extension, so this will be the > line1_name? For example, if I need the extension to be 700 then I set > line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : 701? > > 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP > address of Asterisk, but where? What is the format to write the Asterisk IP > address in this file? > > 4) About the dialplan.xml file, what the below means? > > > > Any help? > Regards > Bilal > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
Hi, OK, many thanks. Got it and I now know the things I need to know. Many thanks, Christian On 2011-06-12 at 17:12 Terry Brummell wrote: >-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian >Sent: Sunday, June 12, 2011 1:51 PM >To: asterisk-users@lists.digium.com >Subject: Re: [asterisk-users] A question about Caller ID > >Hi, >over anlog lines. >Many thanks, >Christian > > > > >Bell 202 modulation between the first and second rings... > >Read the "operation" section from the following link. >http://en.wikipedia.org/wiki/Caller_id > > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files
Hi All; I need to create the needed files for the Cisco Phones to be placed in the TFTP server to be able to register on Asterisk. I need a help in the following please: 1) Regarding to the file: SIPDefault.cnf, The proxy1_address is the IP address of Asterisk? 2) Regarding to the file: SIP.cnf, if I need to let it appear at the Phone (the first line for example) the extension, so this will be the line1_name? For example, if I need the extension to be 700 then I set line1_name : 700 ? And if I need line 2 to be 701 then I set line2_name : 701? 3) About the xmlDefault.CNF.XML, it is mention that I have to place the IP address of Asterisk, but where? What is the format to write the Asterisk IP address in this file? 4) About the dialplan.xml file, what the below means? Any help? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Sent: Sunday, June 12, 2011 1:51 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] A question about Caller ID Hi, over anlog lines. Many thanks, Christian Bell 202 modulation between the first and second rings... Read the "operation" section from the following link. http://en.wikipedia.org/wiki/Caller_id -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco
On Sun, 12 Jun 2011, bilal ghayyad wrote:
How can I see the the files are now loaded? Normally in the windows, I
can see that a request reached to the TFTP server and the files now are
loaded (upload or download)?
The tftp daemon should log the transfers via syslogd. For example:
Jun 12 12:03:23 dt in.tftpd[11486]: RRQ from 192.168.0.19 filename /spa841.cfg
I start tftp via xinetd and the configuration looks like this:
service tftp
{
cps = 100 2
bind= 192.168.0.1
disable = no
flags = IPv4
per_source = 11
protocol= udp
server = /usr/sbin/in.tftpd
server_args = -c -s /tftpboot -v -v -v -v -v -v
socket_type = dgram
user= root
wait= yes
}
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Siemens gigaset as180 as a internal mobile extension
Le 11/06/2011 17:54, Gordon Henderson a écrit : On Sat, 11 Jun 2011, Florent THOMAS wrote: Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. The AS180 is just a bog-standard analogue DECT phone. So like any other analogue phone, to use it with asterisk, you need an analogue card (eg. tdm400p) or an ATA. Personally, I'd use one of the Gigaset IP range of phones - these are SIP compatable. (e.g. A580IP, etc.) They do work very well. I've deployed a fair few in the past few years. (And they're just Gigaset now - split from Siemens AIUI, although I suspect it'll be a long time before everyone catches-up!) Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for answering so fast. I'll try it ASAP and let you know. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
Hi, over anlog lines. Many thanks, Christian On 2011-06-12 at 12:57 Alex Balashov wrote: >Over analog lines? Or ISDN? > >-- >Alex Balashov - Principal >Evariste Systems LLC >260 Peachtree Street NW >Suite 2200 >Atlanta, GA 30303 >Tel: +1-678-954-0670 >Fax: +1-404-961-1892 >Web: http://www.evaristesys.com/ > >On Jun 12, 2011, at 12:42 PM, "Christian" wrote: > >> Hi all, >> Sorry if this is a little off topic, but I just want to know a thing >here. >> What system is used for sending out the caller's number in the US? >> Here in Sweden we use DTMF to send the number out. I just need to know >what is used in the US since I don't think I will be able to use an >American caller ID device over here. >> Many thanks for any info, >> Christian >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco
Dears; Thanks alot for the kindly help. I installed the tftp-server, and I configured it. How can I see the the files are now loaded? Normally in the windows, I can see that a request reached to the TFTP server and the files now are loaded (upload or download)? >From the other side, about the res_phoneprov_and_TFTP, actually I do not see >it that it is used to download the firmware and configuration files, but I see >it is used for provisioning, correct? Regards Bilal - > bilal ghayyad wrote: > > Any one can suggest a TFTP server to be installed in > Fedora > > The one in the repos should be fine. I'm not Fedora > (I'm Mandriva), but > both are RPM based. Try: > > yum install tftp-server. The one that comes with > Mandriva works fine > with our Cisco 7940/7960 phones. > > Doug > --- > > Hi All; > > > > Any one can suggest a TFTP server to be installed in > Fedora (same machine that Asterisk is installed) to be used > for Cisco IP Phones to download the required firmware and > configuration files. > > > > Thanks for the help in advance. > > Regards > > Bilal > > > > Try this > http://etel.wiki.oreilly.com/wiki/index.php/Dynamic_Phone_Provisioning_with_res_phoneprov_and_TFTP > > -- > ~~~ Andrew "lathama" Latham lath...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question about Caller ID
Over analog lines? Or ISDN? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 12, 2011, at 12:42 PM, "Christian" wrote: > Hi all, > Sorry if this is a little off topic, but I just want to know a thing here. > What system is used for sending out the caller's number in the US? > Here in Sweden we use DTMF to send the number out. I just need to know what > is used in the US since I don't think I will be able to use an American > caller ID device over here. > Many thanks for any info, > Christian > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A question about Caller ID
Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LXC and Dahdi
On Sun, 12 Jun 2011, Tzafrir Cohen wrote: On Sun, Jun 12, 2011 at 11:56:52AM +0200, Olivier wrote: Hi, Does it make sense to use LXC as mean to quickly switch from one dahdi version to another or shall other virtualization technologies be preferred ? LXC is basically the same as OpenVZ and Linux-VServer (and probably like Solaris Zones and FreeBSD Jails): The processes of the guest still run inder the same host (specifically: share the same kernel). That's the important bit: One Kernel - that means the version of dahdi you load is shared amongst all containers you use, so trying to get one container run with a different version is not possible. LXC (or rather: cgroups) is now prt of mainline kernel. From palying with it on Debian Squeeze, it feels still a bit immature (and even more so: lacking documentation). But as the usage of cgroups grows in the coming years (e.g.: in systemd), I would expect to see it more and more common. I've been using it for virtual servers for some time now - well, since late last year. Both for Asterisk as as general purpose virtual LAMP hosting. Documentation is an issue, and it's a bit of a steep learning curve and not everything that people post to the various other lists, etc. is relevant to what I want it for, so there's a little bit of "making it up as you go along" - at least for me, but I now have something that works well. In my early experiments, I used an old 1.8Mhz Celeron as a test-bed - I installed 20 containers, each running asterisk and arranged one to call the next to call the next ... to call the first ... and when the last one got to a certian count it connected itself to MoH... I then got a SIP phone to call the first one... Without doing anything clever, I got it to loop 3 times before it showed signs of stress... So so each asterisk only passing media for 3 calls, but that's effectively 60 calls all passing media between them - the kernel time was starting to get high at that point - I reckoned it was probably the network stack was topping out more than anything else. When I did this on production hardware (quad core, 3GHz cpus), that setup was barely noticable. KVM, Xen, VMWare and the likes "emulate" a complete virtual machine. Not just a set of processes. Specifically, your processes run on top of their own kernel. This requires more resources, but provides better isolation. I briefly played with KVM, but you really need a CPUs that support a few extra instructions to make it more efficient - even then, it's not as efficient as LXC is, however it does have other benefits - differenet kernels, different modules, kill a kernel, you don't kill the host, etc. However for my needs, LXC is working very well. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LXC and Dahdi
On Sun, Jun 12, 2011 at 11:56:52AM +0200, Olivier wrote: > Hi, > > Does it make sense to use LXC as mean to quickly switch from one dahdi > version to another or shall other virtualization technologies be preferred ? LXC is basically the same as OpenVZ and Linux-VServer (and probably like Solaris Zones and FreeBSD Jails): The processes of the guest still run inder the same host (specifically: share the same kernel). LXC (or rather: cgroups) is now prt of mainline kernel. From palying with it on Debian Squeeze, it feels still a bit immature (and even more so: lacking documentation). But as the usage of cgroups grows in the coming years (e.g.: in systemd), I would expect to see it more and more common. KVM, Xen, VMWare and the likes "emulate" a complete virtual machine. Not just a set of processes. Specifically, your processes run on top of their own kernel. This requires more resources, but provides better isolation. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LXC and Dahdi
Hi, Does it make sense to use LXC as mean to quickly switch from one dahdi version to another or shall other virtualization technologies be preferred ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
