Re: [asterisk-users] problem transferring calls some of the times
Hi, Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM: Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj No I don't park the call. The call comes in, and gets redirected to our receptionists phone, from there it gets transferred to another extension (the bosses secratary) and then gets transferred (to the boss). now the problem, sometimes that transfer fails, other times the call dont even want to leave the receptionists phone. The big thing about this problem is that it comes and goes, like yesterday we didn't have a problem, and I did not change a thing. Ian Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon: 012 664 2300 Selfoon : 079 522 6519 Faks: 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] IAX: No outgoing audio for 10 seconds
try setting transfer=no or notransfer=yes in iax.conf Depending on the age of your asterisk version. Tim. - Original Message - From: randulo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 20 February 2008 18:12:01 o'clock (GMT) Europe/London Subject: [asterisk-users] IAX: No outgoing audio for 10 seconds I have an IAX hardphone connected to an Asterisk appliance sending and receiving calls via IAX to three different providers. The appliance is currently connected to a NAT router. The appliance is purposely being set up via the GUI, not in messing with any config directly. One of the service providers is my other asterisk box on a different Internet connection. In all cases, outgoing audio is silent for about 10 seconds. If this were SIP, I'd immediately be suspicious of NAT problems, but IAX? Ideas? I will be trying to hang the appliance on the Internet directly but at the moment, this can't happen. -- Join our social networks+voip beta program : http://www.phonefromhere.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen For the brave: use modules.conf without 'autoload = yes'. This promises you many hours of interesting dialplan debugging. Enjoy. Is there any method of automatically parsing a dialplan and generating a list of the modules required to support it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Hi All
Agter a bit of logging to a syslog server, I found a peculiar entry
today, ironically right after a call failed to transfer. They key
sequence and call path used until it gets transferred is as follows
* Phone rings on Asterisk
* Asterisk transferres to the receptionists phone (GXP 2000)
o Receptionist doensnt answer for 15 seconds, and the call
gets routed to the bosses secrataries phone
* Bosses secretary answers the phone and tries to transfer it to the
boss with the keysequence flash, extention 315, talks,
transfer but the transfer is the one that fails
Message in the log is
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] Received SIP message: 407
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] SIP dialog matched to channel 0
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] Send SIP message: ACK To
10.219.127.7:5060, sip_handle: 0x0046F09C
Feb 22 09:55:22 10.219.127.102 GS_LOG:
[00:0B:82:13:02:CF][000][FFFD][01010412] sip_len: 553, sip_handle:
0x0046F09C, ACK sip:[EMAIL PROTECTED];user=p
hone SIP/2.0 Via: SIP/2.0/UDP
10.219.127.102:5060;branch=z9hG4bK623e473ec5e8c5e8 From: Wanda
sip:[EMAIL PROTECTED];user=phone;tag=1a7b934ecd3e23f7 To:
sip:[EMAIL PROTECTED];user=phone;tag=as07aa3c42 Contact:
sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Supported:
path Call-ID: 1138f5f7
[EMAIL PROTECTED] CSeq: 58150 ACK User-Agent: Grandstream
BT200 1.1.4.18 Max-Forwards: 70 Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE,UPDATE,PRACK Content-Length: 0
The message that got my worried is the one saying Recieved SIP message
407, can that be the ghost I am looking for?
Extentions.conf
[incoming_calls]
exten = s,1,NoOp(${CALLERID(name)} skakel Luzaan)
exten = s,n,dial(SIP/300,15)
exten = s,n,Set(CALLERID(name)=deur)
exten = s,n,Set(CALLERID(num)=deur)
exten = s,n,NoOp(${CALLERID(name)} skakel Wanda)
exten = s,n,dial(SIP/312)
;exten = s,n,dial(SIP/317)
exten = s,n,Hangup()
[internal]
exten = 900,1,Verbose(1|Echo test application)
exten = 900,n,Echo()
exten = 900,n,Hangup()
;interne oproepe
exten = _3XX,1,NoOp(${CALLERID} skakel ${EXTEN})
exten = _3XX,n,Dial(SIP/${EXTEN},30)
;exten = _3XX,n,execif(${CALLERID} != _3XX|goto|incoming_calls/s/1)
exten = _3XX,n,goto(incoming_calls,s,1)
exten = _3XX,n,Hangup()
Sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeers=yes
allowtransfer=yes
callevents=yes
regcontext=GXP_BLF
[sets](!)
type=friend
context=internal
host=dynamic
;disallow=all
;allow=speex
secret=test
dtmfmode=info
callgroup=1
pickupgroup=1
call-limit=20
subscribecontext=GXP_BLF
canreinvite=yes
nat=no
[300](sets) ;Luzaan
regexten=300
Any help will be apreciated
Thanks
Ian
Ian said the following on 22-Feb-08 10:06 AM:
Hi,
Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call to 700. No problem. Later, when it's picked up from
701, can it NOT be transferred again?
Moj
No I don't park the call.
The call comes in, and gets redirected to our receptionists phone,
from there it gets transferred to another extension (the bosses
secratary) and then gets transferred (to the boss). now the problem,
sometimes that transfer fails, other times the call dont even want to
leave the receptionists phone.
The big thing about this problem is that it comes and goes, like
yesterday we didn't have a problem, and I did not change a thing.
Ian
Ian wrote:
Hi All
Sorry to be a bother again but seems like I just cant get away from
the problems.
This time my problem is that *sometimes* a user cant transfer a call
from one extension to another, I have narrowed down the problem to it
only happening to calls from outside the internal system.
The wierd thing about the problem is that it comes and goes one moment
the user can transfer, and the next call he can't.
I am running:
* Asterisk 1.4.17
* Zaptel 1.4.7.1
* Libpri 1.4.3
Using the following phones and firmware
* Grandstream GXP2000 (with ext pad) : 1.1.4.14
* Grandstream BT200 : 1.1.4.18
I have set up the phones to log debug logs to a syslog server, I am
still trying to figure out what exactly the log says.
Is it an * problem, or Grandstream problem
Does anyone know if I am able to see the keysequence the user types
into the phone (just in case it might even be a user made problem), I
have tried scanning though the logs of a failed call, but could not
see any lines that can be a keypress, or maybe I am looking in the
incorrect spot?
Your help will be greatly appreciated.
Let me know if, in any way, I can shed some more light on the subject.
Thanks in advance
Ian
--
www.vddi.co.za http://www.vddi.co.za/
I Coetzee
IT Tegnikus
Telefoon: 012 664 2300
[asterisk-users] Weird Zaptel sound after anwser calls
Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a Clack. I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes callerid=asreceived transfer=yes callreturn=yes rxgain=-3.0 txgain=-3.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both group=0 signalling=fxs_ks channel = 7 think the problem is not by echo cause I use fxotune, and the problem persist. I made lots of TDM02B instalations and never get this kind of problem. Any clue will be welcomed. Thanks in advance. Regards VoipCrazy ___ -- 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, Zaptel and the Kernal Compatibility Matrix
On Fri, Feb 22, 2008 at 02:33:01AM -0500, Matt Florell wrote: Hello, I was never able to get the TE407P card running on a 2.4 Linux kernel. Using a 2.6 kernel I was able to get it working. What error(s) do you get? On what platform / kernel exactly? Not really surprising since a lot of companies do not support or even test on Linux 2.4 any more. Well, nobody really likes 2.4, I guess. Build logs of zaptel svn on various kernels from recent days. A certain 2.4 is included there: http://updates.xorcom.com/logs/zaptel/report.html -- 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] How to get a clean, basic configuration?
On Fri, Feb 22, 2008 at 01:28:58AM -0800, Steve Langstaff wrote: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen For the brave: use modules.conf without 'autoload = yes'. This promises you many hours of interesting dialplan debugging. Enjoy. Is there any method of automatically parsing a dialplan and generating a list of the modules required to support it? You'd have to also know which modules were loaded and what applications they have registered. I suspect doing so would take a partial loding of Asterisk. How is apache's configtest implemented? -- 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] Weird Zaptel sound after anwser calls
I forgot to say that I'm using bristuff-0.4.0 with zaptel 1.4.4, libpri 1.4.1 and asterisk 1.4.9 Thanks. 2008/2/22, voip crazy [EMAIL PROTECTED]: Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a Clack. I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes callerid=asreceived transfer=yes callreturn=yes rxgain=-3.0 txgain=-3.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both group=0 signalling=fxs_ks channel = 7 think the problem is not by echo cause I use fxotune, and the problem persist. I made lots of TDM02B instalations and never get this kind of problem. Any clue will be welcomed. Thanks in advance. Regards VoipCrazy ___ -- 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] Using app_sms in South Africa
Anyone from South Africa out there that has gotten SMS over Telkom lines right? Ive found the SMSC but I dont have the foggiest how to go about it ___ -- 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] Voted most stable and easy to use phone?
Linksys SPA942. Tried most of available phones on the market. These phones sits on companies tables for more then a year. No problem at all, easy to use, nice(!) to use. I recommend to everybody. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, February 22, 2008 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voted most stable and easy to use phone? On Thu, Feb 21, 2008 at 7:32 PM, arkda [EMAIL PROTECTED] wrote: I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are pretty straightforward to manage via TFTP, and work really well with Asterisk. I agree, we've had zero trouble with these. Easy to install and they just work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Digium B410P and 8 ports connectivity
Hello, Junghanns and BeroNet offer 8 BRI ports cards. Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out of 2 B410P ? I know you could (theorically) do this with Junghanns and BeroNet cards. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote: For the brave: use modules.conf without 'autoload = yes'. This promises you many hours of interesting dialplan debugging. Enjoy. Yup, that's what I anticipated, which is why I was asking which modules I can _safely_ remove without breaking things :-) ___ -- 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] canreinvite question
Hi All, if i do this setup: |---[ext 100] |--[router/nat gw]--| | |---[ext 101] | [asterisk]--[internet]---| | | |---[ext 200] |--[router/nat gw]--| |---[ext 201] If i set, canreinvite=yes on all ext, assuming all ip phones have the same codec, if 100 calls 101, or vice versa will rtp still go thru asterisk? and same scenario for 200 to 202 or vice versa. what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go thru asterisk? thank you regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
On Thu, 21 Feb 2008 08:33:20 -0500, C F [EMAIL PROTECTED] wrote: first off I anwered you to use vi and you complained showing me cat. There's some misunderstanding. I didn't complain. I just didn't know if Asterisk only looked for stuff in modules.conf because there was so little there and so much stuff scrolling by when I type reload. Before loading modules explicitely, I need to make sure what each does precisely, and what the interdependencies are, if any, so that I know what the consequences are if I decide not to load a mdoule that looks like it's not needed on my setup. 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] Pattern matching....
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote: No that will not work. You would want three exten = lines, one for each area code. And if you have a lot of common dialplan that you don't want to duplicate between the three extension patterns, put the common stuff up at a higher priority and use Goto to get there: exten = _404NXX,1,Goto(200) exten = _770NXX,1,Goto(200) exten = _678NXX,1,Goto(200) exten = _NXXNXX,200,NoOp(Start of common instructions) exten = _NXXNXX,n,etc Cheers Tony Michael Munger wrote: Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Attachment encrypted? click here http://www.highpoweredhelp.com/tutorials/wincrypt/ . ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite question
On Fri, 22 Feb 2008 18:50:16 +0800, Ron [EMAIL PROTECTED] wrote: If i set, canreinvite=yes on all ext, assuming all ip phones have the same codec, if 100 calls 101, or vice versa will rtp still go thru asterisk? and same scenario for 200 to 202 or vice versa. ... and I'd like to add to this question: If the phones have the option Enable NAT, I expected them to be able to talk to each other directly, but they didn't, and I had to set them to canreinvite=no in sip.conf. Why is that? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voted most stable and easy to use phone?
They have their ups and downs. If you live outside the US, localising your tones is a pain in the proverbial since you have to define every tone by frequency combination and intervals, although I guess you do only need to do it once. One other shortcoming of the 942 is the lack of any usable BLF buttons at all. I'm led to believe that one of the upcoming firmware releases will allow you to use the four line buttons as BLF buttons, but since you need at least two line buttons in order to be able to transfer a call, that means you realistically would only get two BLF keys - not many - especially if you're used to the Snoms or Grandstreams (hack, spit) The 962s are /really/ nice phones, albeit somewhat more expensive. The large colour display is /very/ nice, and button banks for the 962s are available and very cheap compared to many other options, even if they do look a little cheap. arkda wrote: I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are pretty straightforward to manage via TFTP, and work really well with Asterisk. On Thu, Feb 21, 2008 at 4:07 AM, Michael J. Liberatore [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. Basically i am looking to get some new phones and in the process get rid of the countless number of problems i have had that has always been caused by phones (snom 360's and gxp-2000's). I would like to get the feedback of the list on the phone voted best for stability, working with *, and ease of use for dumb non tech users. I was thinking of trying one of these new polycom phones that are about $150, but havent gotten any feedback on them yet. Basically i am interested in any phones but snom's, grandstreams, and sipura's/linksys. mainly polycom's i guess. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern matching....
That's what I figured, and the way I've always done it. I was just *hoping* someone knew of a better way that I didn't know about. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Thursday, February 21, 2008 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pattern matching On Thu, 2008-02-21 at 13:34 -0500, Mike Trest - Personal wrote: [746][704][048] Nope, that's not going to do exactly what you want either... that pattern would match a lot of area codes besides the ones you're looking for. (For example, you could have 7 as the first digit and 0 as the second digit and 0 as the third digit.) You really need to have three separate patters; one for each area code. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Linksys SPA-942 Phones
Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: * How do the phones handling system wide paging? Is it similar to the Polycom phones? * Can a corporate directory be configured with the phones using Asterisk? * How is the speaker phone quality? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix
Hello, This was about a year ago when we abandoned putting new systems on the 2.4.33 kernel. About that time we started having other vendors stop supporting it very well also so it wasn't only that card that we were having issues with. The problems were related to the CRC Linux modules and zaptel not liking them in 2.4. One very interesting side note, on identical hardware we got 5-10% better Asterisk performance on a 2.4 kernel than we got on 2.6 kernels. Overall we are happy with 2.6 and it's new features and we don't really want to go back at this point. I just wanted to mention our experiences with kernel problems. Thanks, MATT--- On 2/22/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 02:33:01AM -0500, Matt Florell wrote: Hello, I was never able to get the TE407P card running on a 2.4 Linux kernel. Using a 2.6 kernel I was able to get it working. What error(s) do you get? On what platform / kernel exactly? Not really surprising since a lot of companies do not support or even test on Linux 2.4 any more. Well, nobody really likes 2.4, I guess. Build logs of zaptel svn on various kernels from recent days. A certain 2.4 is included there: http://updates.xorcom.com/logs/zaptel/report.html -- 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 ___ -- 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] DUNDi ${NUMBER} variable not defined
If one does a dundi lookup, shouldn't the ${NUMBER}
variable be replaced with the current value?
ie. if I run dundi lookup [EMAIL PROTECTED] shouldn't I get
an answer string like
IAX2/priv:[EMAIL PROTECTED]/4065 (EXISTS)?
The *CLI does not show me the dst extension:
*CLI dundi lookup [EMAIL PROTECTED]
1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 53 ms
*CLI dundi lookup [EMAIL PROTECTED]
1. 0
IAX2/priv:[EMAIL PROTECTED]/${NUMBER}
(EXISTS)
from 00:1d:60:b0:25:10, expires in 5 s
DUNDi lookup completed in 37 ms
What could be wrong?
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
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[asterisk-users] Opinions please: Polycom IP 430 vs 330?
I need to add a few phones to an existing installation. They have a dozen IP430 at the moment. Does anyone feel that there are advantages to the IP330? Cost is not the major consideration as long as they're in the same range. (under $175) Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?
Let me add another variable into the mix...what about the Linksys SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a deal at $80 street price. Michael On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote: Michael Graves wrote: I need to add a few phones to an existing installation. They have a dozen IP430 at the moment. Does anyone feel that there are advantages to the IP330? Cost is not the major consideration as long as they're in the same range. (under $175) Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 The 430 has a larger footprint and feels to be a slightly heftier and more business-like phone. I have very similar sound quality and results with either phone. To me it would be a toss-up based upon the customer's taste. -- Join the FSF as an Associate Member at: URL:http://www.fsf.org/register_form?referrer=5175 --~--~-~--~~~---~--~~ Your participation in the conference is always appreciated! Please try to be there live when it happens. You received this message because you are subscribed to the Google Groups Asterisk Users Conference group. To post to this group, send email to [EMAIL PROTECTED] To unsubscribe from this group, send email to [EMAIL PROTECTED] For more options, visit this group at http://groups.google.com/group/VOIP-Users-Conference?hl=en -~--~~~~--~~--~--~--- -- Michael Graves mgravesatmstvp.com http://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] Will this be sufficient for 20+ concurrent calls?
This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333 Mhz. 2 GB RAM, 3.5 SATA II discs. I know this is a rather vague question that can not be precisely answered, but mainly handling calls w/ playback of local gsm-files - that is, no redirection to other clients etc. - will it be sufficient for 20-something concurrent calls? ___ -- 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] Will this be sufficient for 20+ concurrent calls?
More than enough. Julian. harry wrote: This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333 Mhz. 2 GB RAM, 3.5 SATA II discs. I know this is a rather vague question that can not be precisely answered, but mainly handling calls w/ playback of local gsm-files - that is, no redirection to other clients etc. - will it be sufficient for 20-something concurrent calls? ___ -- 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 __ This email for dotr.com has been scanned by MessageLabs __ ___ -- 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] adjusting volume on a wildcard 100XP with zaptel's {t, r}xgain
I'm finding the volume of the calls on my wildcard 100XP (clone) is too low. I understand I can muck with rxgain and/or txgain (which one in fact will increase the volume of the other party as far as I'm hearing it?) to deal with this but right now I have them both at 0.00 and I am concerned about introducing echo by fiddling with them given that these two registers are what are fiddled with to deal with echo problems too. How can I safely adjust these? Caller report low volume problems from me too, so if I make equal adjustments to both will that maintain my level of (non-)echo? Thanx, b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High CPU load after upgrading to 1.4
I verified that qualify=no. I am getting this CPU load at 10% without any peers even registered which is very strange, but it doesn't happen when I run on 64-bit CentOS 5 kernel. Remi - Original Message - From: Jared Smith Date: Thursday, February 21, 2008 5:55 pm Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4 To: Asterisk Users Mailing List - Non-Commercial Discussion On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote: I currently have 1558 sip peers loaded in Asterisk and the current CPU load is 10% when no calls are being processed and no sip registrations. At first glance, I would think that maybe you have qualify=yes in each of your SIP peers, which is keeping Asterisk busy checking to see if the peers are responding or not. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Will this be sufficient for 20+ concurrent calls?
Harry, I think this system will suffice for your needs. I have a similar setup working great with 2 Dual Core Xeon @ 2GHz On Sat, Feb 23, 2008 at 9:21 AM, harry [EMAIL PROTECTED] wrote: This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333 Mhz. 2 GB RAM, 3.5 SATA II discs. I know this is a rather vague question that can not be precisely answered, but mainly handling calls w/ playback of local gsm-files - that is, no redirection to other clients etc. - will it be sufficient for 20-something concurrent calls? ___ -- 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 -- Nacho Linux Counter #156439 ___ -- 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] Interrupt VM and Steal a call.
Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're leaving it? 2. If a phone rings a receptionist desk, and the receptionist is down the hall, she wants to be able to dial an extension, and have that transfer the call from her desk to the phone she's currently on so she doesn't have to run to her desk. Is there a built in feature for this or do I have to code it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Attachment encrypted? click here http://www.highpoweredhelp.com/tutorials/wincrypt/ . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to the voicemail of calle's extension.
It's really pretty easy.
; Call the SIP peer, let the phone ring for 20 seconds
exten = 123,1,Dial(SIP/some_sip_peer,20)
; Play the press-1-or-press-2 prompt, get one digit
; from the caller, and save it to a variable called
; ${option}
exten = 123,n,Read(option,press-1-or-press-2,1)
; If the caller enters 1, send the call to the [some_context] context,
; to the operator extension, priority 1
exten = 123,n,GotoIf($[${option} = 1]?some_context,operator,1)
; Otherwise, send the call to voicemail
exten = 123,n,VoiceMail([EMAIL PROTECTED])
I haven't actually taken the time to test this in my own dialplan, but
it should work. Obviously you'll want to change the name of the SIP
peer you're dialing, as well as the location of the operator extension.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Will this be sufficient for 20+ concurrent calls?
On Fri, 2008-02-22 at 14:51 +0100, harry wrote: This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333 Mhz. 2 GB RAM, 3.5 SATA II discs. I know this is a rather vague question that can not be precisely answered, but mainly handling calls w/ playback of local gsm-files - that is, no redirection to other clients etc. - will it be sufficient for 20-something concurrent calls? That server should be more than adequate for 20 concurrent calls. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?
On Fri, 2008-02-22 at 07:43 -0600, Michael Graves wrote: Let me add another variable into the mix...what about the Linksys SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a deal at $80 street price. I'm quite happy with my SPA-962 + sidecar... I tend to use it more than the other phones I have here in my hardware zoo. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support 20 user's conference?
On Thu, 2008-02-21 at 13:57 +0800, zhao_x_q wrote: HI, Friends, Now I have 20 polycom’s SS2 phones. Can Asterisk support 20 users conference meeting? Yes. And I want to build HD audio conference by using polycom’s 650 ip phone. Can asterisk support G722 HD audio conference? Afaik Asterisk only supports it in 1.6beta. If you need a working solution *now* then have a look at FreeSWITCH which supports wideband and ultra-wideband conferences very well. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which codec over iax = pstn
Atis Lezdins wrote: BTW, we have 512kbs over the iax connection. G711 needs about 80Kb/sec each way to work. (It's 64Kb/sec plus IP overhead). GSM needs about 32Kb/sec (13Kb/sec plus IP overhead). So with DSL 512kbs up and 3mbs down, plenty of room for G711. Take the weakest link - up 512 kbps, that makes 6 simultenous ulaw calls (not counting other traffic). Of course you could push more calls, inbound voice would be good (3mbps) but outbound would be crappy. Of course, I could figure out how to configure QOS in iptables for asterisk, it'd be a lot better. If you have fixed bandwidth, it should be fairly simple, there's some ready scripts for scheduling outbound/inbound traffic on fixed bandwidth links. This is a very good resource for that - http://lartc.org/ Also i found this yesterday, could be good for start. It doesn't assume fixed bandwidth, but just gives priority to VoIP packets. http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk That's a great find. I've never been able to figure out the lartc docs. This really lays it out, simply. Should be linked from the digium or voip-info sites. Regards, Atis ___ -- 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] Digium B410P and 8 ports connectivity
Olivier wrote: Hello, Junghanns and BeroNet offer 8 BRI ports cards. Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out of 2 B410P ? You can also use the Sangoma A500 for up to 24 BRI ports. You get 6 ports on the main card and 6 on every remora. Only 1 PCI slot is used. The remoras don't require the slot, just the physical space. Andres. http://www.neuroredes.com I know you could (theorically) do this with Junghanns and BeroNet cards. Cheers ___ -- 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] Can asterisk support 20 user's conference?
On Fri, Feb 22, 2008 at 03:22:39PM +0100, Patrick wrote: And I want to build HD audio conference by using polycom’s 650 ip phone. Can asterisk support G722 HD audio conference? Afaik Asterisk only supports it in 1.6beta. If you need a working solution *now* then have a look at FreeSWITCH which supports wideband and ultra-wideband conferences very well. Both Asterisk 1.6 and FreeSwitch are not officially a stable release. -- 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, Zaptel and the Kernal Compatibility Matrix
On Fri, Feb 22, 2008 at 07:59:14AM -0500, Matt Florell wrote: Hello, This was about a year ago when we abandoned putting new systems on the 2.4.33 kernel. About that time we started having other vendors stop supporting it very well also so it wasn't only that card that we were having issues with. The problems were related to the CRC Linux modules and zaptel not liking them in 2.4. Try again, then. In the recent monthes there were some 2.4-specific compilation fixes. One very interesting side note, on identical hardware we got 5-10% better Asterisk performance on a 2.4 kernel than we got on 2.6 kernels. Interesting regression. Zaptel is generally a program that stresses some parts of the ernel in unexpected ways. Overall we are happy with 2.6 and it's new features and we don't really want to go back at this point. I just wanted to mention our experiences with kernel problems. Good and detailed bug reports are always welcomed. -- 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] How to get a clean, basic configuration?
On Friday 22 February 2008 04:55:13 Vincent wrote: On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote: For the brave: use modules.conf without 'autoload = yes'. This promises you many hours of interesting dialplan debugging. Enjoy. Yup, that's what I anticipated, which is why I was asking which modules I can _safely_ remove without breaking things :-) Generally, the rule is that you can't remove any of the res_* modules. If you follow that rule, you can then start to work out the applications, functions, and channels that you don't want. The codecs and formats modules are usually too small to consider disabling, unless it is absolutely critical for space (such as on embedded systems, where you generally don't want any transcoding anyway). -- Tilghman ___ -- 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] Interrupt VM and Steal a call.
Michael Munger wrote: Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let’s say the phone goes to voicemail after 4 rings, yet I don’t hear it until the 3^rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they’re leaving it? This can be done with FOP (Flash Operator Panel) or any application that can issue Manager commands to Asterisk. 2. If a phone rings a receptionist desk, and the receptionist is down the hall, she wants to be able to dial an extension, and have that transfer the call from her desk to the phone she’s currently on so she doesn’t have to run to her desk. Is there a built in feature for this or do I have to code it? Yes, this option is accessed via the features.conf file. It's the pickupexten feature. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interrupt VM and Steal a call.
On Fri, Feb 22, 2008 at 09:05:17AM -0500, Michael Munger wrote: Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're leaving it? I imagine that would be tricky to do once the call has been handed in to the voicemail application, as presently you are limited in what you can do to the call once it's gone in there. You might be able to locate the original SIP channel and bridge the calls, but I've no idea how you would track that properly. There is a way to make voicemail have a press 0 to be transferred to somewhere else option. We use that here and it works. Users can set up where they want the caller to be transferred to (usually a mobile) and then they can record on their outgoing message leave me a message, or press 0 to try my mobile... Or, Sounds like a case for a few IP DECT cordless handsets to save all this running about! You might run into somebody carrying a boiling hot cup of coffee in your rush to answer the phone! (happens!) We have a few Siemens C460 IP DECT Phones. The range and battery life on them is by far superior to any of the WiFi/SIP phones I've tried so far. I have a SIP/Wifi Nokia E65 that works great, but the battery life is not very good when the wifi is left on, and it was less than straightforward to set up! 2. If a phone rings a receptionist desk, and the receptionist is down the hall, she wants to be able to dial an extension, and have that transfer the call from her desk to the phone she's currently on so she doesn't have to run to her desk. Is there a built in feature for this or do I have to code it? There is a feature called pickup defined in features.conf: pickupexten = *8 Restart asterisk if you need to change features.conf (in my experience just a reload when changing features.conf doesn't always work) You then need to define your SIP/devices into pickup groups in sip.conf, for example:- [500] canreinvite=yes nat=no secret=... dtmfmode=rfc2833 callgroup=1 pickupgroup=1 [501] canreinvite=yes nat=no secret=... dtmfmode=rfc2833 callgroup=1 pickupgroup=1 Then reload. Now, if extn 500 were ringing, picking up 501 and doing *8 will connect that call. For busier systems I believe there is a dialplan feature that enables directed pickup so you can pick up a specific extension, but I haven't played with that so I can't say how it works. That might be more suitable. The callgroup defines what pickup group the device is in, and pickupgroup defines what groups (when that extension dials *8) that device can pick up. A device can be in one callgroup but multiple pickup groups:- pickupgroup=1,2 This is so that if you have many sites or departments, only people who sit within the range of the ringing phone can pick it up, and not get connected to some other random call incoming somewhere else. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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] High CPU load after upgrading to 1.4
Did you file a bug report? http://bugs.digium.com -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Thursday, February 21, 2008 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4 On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote: I currently have 1558 sip peers loaded in Asterisk and the current CPU load is 10% when no calls are being processed and no sip registrations. At first glance, I would think that maybe you have qualify=yes in each of your SIP peers, which is keeping Asterisk busy checking to see if the peers are responding or not. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk config file online editor
With some mods it surely did the trick
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan
Company, LLC
Sent: miércoles, 20 de febrero de 2008 01:49 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
h1?=$fn?/h1
textarea name=contents?php include $fn ?/textarea
input type=hidden name=action value=write
input type=submit value=Save File input type=reset value=Reset
/form
Security holes galore! clean it up a bit :) And check on permissions
issues, that your httpd can write to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an
interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of IntruderEnterprises S.A. de C.V.
www.Intruder.com.mx
www.IntruderStore.com.mx
Tel. 3872-2200 ext. 201
Tel. 01-800-INTRUDER (01-800-468-7833)
Email: [EMAIL PROTECTED]
Como lo estoy haciendo? Contacte a mi Director:
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Re: [asterisk-users] Voted most stable and easy to use phone?
I guess someone has to say it. Have you considered Aastra? You can argue about quality/features/functionality but I have set up both and the Aastra are definitely easier to configure and they reboot quicker. Nobody ever complains about the quality of sound or speakerphone on them either. From: John Faubion [mailto:[EMAIL PROTECTED] Sent: Thursday, February 21, 2008 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voted most stable and easy to use phone? A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. We just installed a dozen of the Polycom IP-330 phones. Initially out of the box I wasn't real sure about the decision to use them. The phones are very small and don't seem to have very many features. However in use they have been great. They don't waste a lot of desk space, they don't overwhelm the users and they seem to provide adequate information. They're easy to use and Polycom reliable. The speaker phone is still really good though I'm not sure it is as good as the 501/601 phones. I haven't really done a side by side comparison of that but I think the 501/601 has a better speaker phone. I can't see buying another GXP after using these. The difference in price just isn't worth the aggravation. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
vi /etc/asterisk/extensions.conf On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote: hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep. ___ -- 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] problem transferring calls some of the times
Are you using buttons on your phone to effect the transfer, or are you using codes defined in features.conf? Moj Ian wrote: Hi, Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM: Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj No I don't park the call. The call comes in, and gets redirected to our receptionists phone, from there it gets transferred to another extension (the bosses secratary) and then gets transferred (to the boss). now the problem, sometimes that transfer fails, other times the call dont even want to leave the receptionists phone. The big thing about this problem is that it comes and goes, like yesterday we didn't have a problem, and I did not change a thing. Ian Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon: 012 664 2300 Selfoon : 079 522 6519 Faks: 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-942 Phones
On Fri, Feb 22, 2008 at 7:56 AM, Anciso, Roy [EMAIL PROTECTED] wrote: Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: · How do the phones handling system wide paging? Is it similar to the Polycom phones? Yes it's quite similar, just be aware that unlike the Polycoms, if you page a phone (auto answer header) that is in a conversation the SPAs will put the current call on hold and take the auto answered phone, therefore before paging an SPA make sure to use app_chanisavail first. · Can a corporate directory be configured with the phones using Asterisk? Don't know, I never tried it, but IIRC the web page to the SPA configs has a tab for that. · How is the speaker phone quality? For the price it's great, but it's not the best, in my opinion it's acceptable quality. It's not as good as the Cisco which is worst than the Polycoms. I would say it's around as good as the Aastras Thanks Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there ... I've tried doing something similar and came with two options. The common to them is that I use MySQL for realtime extensions, and set systemname parameter to the IP address of the server where the phone registers. When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. I had this working in the lab but not in production yet. If you want the dialplan code for this then email me. __Yehavi: ___ -- 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] load balancing SIP extensions
--- Andres Jimenez [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Sorry, I jut got your other message stating the steps your boss' secretary uses to transfer calls, so this question's time is past. I'm curious if the 'flash' button is the only way those phones can do a transfer. Do they have any other transfer keys, or could you try the featuremap codes? Our polycom transfer buttons have always just worked, but my users, for some reason, all felt more comfortable using DTMF keypresses... dunno why :) So we all press ## to do a blind transfer now, or ** to auto-park to first parking space. Moj Mojo with Horan Company, LLC wrote: Are you using buttons on your phone to effect the transfer, or are you using codes defined in features.conf? Moj Ian wrote: Hi, Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM: Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj No I don't park the call. The call comes in, and gets redirected to our receptionists phone, from there it gets transferred to another extension (the bosses secratary) and then gets transferred (to the boss). now the problem, sometimes that transfer fails, other times the call dont even want to leave the receptionists phone. The big thing about this problem is that it comes and goes, like yesterday we didn't have a problem, and I did not change a thing. Ian Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon : 012 664 2300 Selfoon: 079 522 6519 Faks : 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] FXO Cards - T38
Hi, Could some one let me know if a fax is received through a FXO card connected to * and fax machine is connected to a Linksys FXS device which support T38, is T38 going to be used for faxes which comes from PSTN or go through PSTN ? or because of Asterisk T38 passthrough support it is not possible ? so is for this scenery better to use external FXO gateways with t38 support ? Regards, Fernando ___ -- 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] NOKIA E series Phone for SIP-VOIP calling
Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web also from Nokia site but they only mention this features as VOIP call from wifi they mentioed only this much info. they not mentioed info about inbulit SIP client to make voip calls without download any third party software. as per my search i found this 3 phones E51,E61i,E65 So Guys plz help me for this. Regards Amit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI / Voicemail Que
Hello All,
I have my own AGI script running and I am trying to push the call to
voice mail when Busy, Unavailable and Not Answered.
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only Temp Greetings
voice mail greetings is played. I am passing the correct parameters for
Busy = 'b', Unavailable = 'u' and default goes to Not Answered.
Here is a code sample: -
?
if($status == BUSY){
$arr = array([EMAIL PROTECTED], 'b');
$agi-exec(VOICEMAIL, $arr);
} elseif ($status == CHANUNAVAIL){
$arr = array([EMAIL PROTECTED], 'u');
$agi-exec(VOICEMAIL, $arr);
} else {
$arr = array([EMAIL PROTECTED]);
$agi-exec(VOICEMAIL, $arr);
}
?
Here is the AGI Debug message: -
-- AGI Script Executing Application: (VOICEMAIL) Options:
([EMAIL PROTECTED]|b)
-- Playing '/var/spool/asterisk/voicemail/default/2481237766/temp'
(language 'en')
As you can see I am passing the correct parameter for BUSY = |b, but
Asterisk is only playing the temporary greetings.
Any suggestions... By the way I am running Asterisk 1.2.18
Cheers,
Nitesh
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Re: [asterisk-users] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: --- Andres Jimenez [EMAIL PROTECTED] wrote: On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote: However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On 2/22/08, amit salunkhe [EMAIL PROTECTED] wrote: Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web also from Nokia site but they only mention this features as VOIP call from wifi they mentioed only this much info. they not mentioed info about inbulit SIP client to make voip calls without download any third party software. as per my search i found this 3 phones E51,E61i,E65 I have an E65 and it works quite well with my Asterisk install. No extra software needed, everything is built into those phones. The menus aren't exactly intuitive to get it setup the first time and roaming between access points could be better. All in all I'm happy though. If you do get an E65, make sure you update your firmware to 2.0633.65.01 or greater. It solves a large number of SIP related problems. -Michael ___ -- 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] load balancing SIP extensions
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote: Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. You are right. That could happen if the phone is not registered anywhere You can put some security in the dialplan. if calls comes from IAX it means that PHONE is not registered in the other server. Just create special extensions to take the IAX calls (instead of GoTo): PHONE is 101 SERVER 1 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER2/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup SERVER 2 exten = 101,1, Dial SIP/101 exten = 101,1, Dial IAX-SERVER1/55101 exten = 55101,1, Dial SIP/101 exten = 55101,1, Hangup I hope it helps, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI / Voicemail Que
Nitesh Divecha wrote: ([EMAIL PROTECTED]|b) Any suggestions... By the way I am running Asterisk 1.2.18 I believe under 1.2.x it would be [EMAIL PROTECTED] One of my older dial plans lists: s-BUSY,1,Voicemail([EMAIL PROTECTED]) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On Sat, 2008-02-23 at 00:03 +0530, amit salunkhe wrote: i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web also from Nokia site but they only mention this features as VOIP call from wifi they mentioed only this much info. they not mentioed info about inbulit SIP client to make voip calls without download any third party software. as per my search i found this 3 phones E51,E61i,E65 I've one nokia E65 that works very well with my asterisk box. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- 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] NOKIA E series Phone for SIP-VOIP calling
On Sat, 23 Feb 2008, amit salunkhe wrote: Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web also from Nokia site but they only mention this features as VOIP call from wifi they mentioed only this much info. they not mentioed info about inbulit SIP client to make voip calls without download any third party software. as per my search i found this 3 phones E51,E61i,E65 I have an E90 communicator and it's working just fine. However, when I first got it, it would crash on an incoming call, however it stopped crashing after a few rboots and has been OK since then. A friend also has an E90, and it hasn't stopped crashing on incoming calls. I think they still have a software rev. or 2 to go before it's going to be solid... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Message waiting light on polycom 301 using asterisk 1.4.14
All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I see this on the console chan_sip.c:14907 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 6000 users.conf [6000] fullname = Joe User email = [EMAIL PROTECTED] secret = 1234 ;zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no context = default callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes Thanks for any help you can provide. Richard -- This message was scanned by ESVA and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is tos=ef same as tos=0xb8 same as DSCP ef ?
Trying to figure out how to prefer voip traffic on a dsl line. Found a great howto: http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk but I'm trying to figure out the relationship between the tos of iax.conf and tos of tc from Iproute2. my traffic goes from my linux router to a CPE cisco box. I understand Cisco uses tos ( usually referred to as DSCP, just to keep us on our toes) ef or 0xb8. So when I use tos=ef in *.conf, will the Cisco router see it as Cisco's ef, DSCP=ef? And, for extra credit, can I use tos 0xb8 in iproute2: tc filter add dev eth1 protocol ip parent 1: prio 1 u32 match ip tos 0xb8 0xff flowid 1:2 It looks to me that priomap only has 16 values, so anything over 0x10 won't work. Thanks sean ___ -- 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] Linksys SPA-942 Phones
* How do the phones handling system wide paging? Is it similar to the Polycom phones? No idea I'm afraid, none of our clients use paging functionality. * Can a corporate directory be configured with the phones using Asterisk? Yes and no. You can set up a directory on a per-phone basis through the web interface, but it isn't overwritten by a provisioning config. You can fudge the issue some with curl and a php (or insert choice of preferred scripting language) script. * How is the speaker phone quality? Excellent. Noticably better than either the Snom 370 or Snom 300 that also populate my desk. Polycoms are a bit harder to come by here in the UK, so I haven't had much experience with them and can't do a direct comparison, but definitely much better than the speakerphones in the Snom units. I would say on a par with the Cisco 7960 (which also graces my desk). Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message waiting light on polycom 301 using asterisk 1.4.14
Cavanna, Richard wrote: All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I see this on the console chan_sip.c:14907 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 6000 users.conf [6000] You're missing: [EMAIL PROTECTED] You'll also need to edit the voicemail.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 301/501 Keymapping
I know how to remap a key on a polycom 301 and 501 But does anyone know of a list of mapping keys? For example, the Do Not Disturb on a 301 is #23. I got that one by just guessing though. Thanks, Rob ___ -- 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] cid_rewrite.php -- Caller ID Name lookup
Jay Milk wrote: For those folks who are still using it -- I updated the cid_rewrite script. I noticed that two of the providers were iffy and one had changed format a little while ago. It's working again. http://muware.com/asterisk has the latest (1.2.0) Updated to 1.2.1 to fix an issue with how test_cid.php received parameters. Thanks to the friendly soul for pointing this out. ___ -- 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] Digium B410P and 8 ports connectivity
Olivier wrote: Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out of 2 B410P ? No, the card does not support that mode. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Post call QoS in Asterisk 1.4
It's time to ask this question again. Maybe I will get a reply one day. :) Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] 1.4 and IAX Trunks ...
Gordon Henderson wrote: What about the need for 1.4 at all sites? Is it sufficient to just have it in the man in the middle site? It uses new IAX2 commands, so it requires that all three endpoints understand those commands. At this time the only IAX2 implementations that I am aware of that support media-only transfers are Asterisk 1.4 and Asterisk 1.6 beta releases. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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] spandsp/tx_fax/rx_fax frustrations
Edwin, I feel your pain. I struggled getting fax to work reliably with both 1.2 and 1.4 versions. Any combination I tried, usually caused a crash. I recently upgraded to 1.6.beta4 and installed the app_fax from the addons installation and it worked first time out of the box :-) I've received a couple dozen test faxes without any crashes. I realize it doesn't help your 1.2 needs, but I can confirm your frustration. Jim Edwin Lam wrote: hi does any body know which version combination of spandsp/tx_fax/rx_fax will work with * 1.2.24? i tried different combo. they're either seg fault during runtime or won't compile. very frustrated :/ p.s. i know. hylafax/iaxmodem is far more stable. but i have specific reasons to use rx_fax. ___ -- 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] Post call QoS in Asterisk 1.4
I have absolutely no idea since I was not even aware of it. However, this may give you some hints as to where you can find more information: http://www.mail-archive.com/[EMAIL PROTECTED]/msg27124.html - Waldo On Feb 22, 2008, at 5:08 PM, Douglas Garstang wrote: It's time to ask this question again. Maybe I will get a reply one day. :) Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your homepage. ___ -- 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] Polycom 301/501 Keymapping
That can be found in the monstrous admin guide for the phone, seemly in Section 3.1.7 in my ancient version 1.5.0 document. It shows me that on the 501, that button is 9 instead of 23. http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html There's a link to the administrator's guide down there under Setup Maintenance Documents. Moj Rob Schall wrote: I know how to remap a key on a polycom 301 and 501 But does anyone know of a list of mapping keys? For example, the Do Not Disturb on a 301 is #23. I got that one by just guessing though. Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mexico Dids
Hi, I am looking for a did from Saltillo Mexico. Any pointers? robert ___ -- 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] Linksys SPA-942 Phones
So far I've never run into anything that's even /close/ to the speakerphone quality of the Polycoms. There's no comparison on the speakerphone between the Linksys phones and the Polycoms - it's chalk and cheese, but by the same token that holds true for just about every other phone too. Chris Bagnall wrote: * How is the speaker phone quality? Excellent. Noticably better than either the Snom 370 or Snom 300 that also populate my desk. Polycoms are a bit harder to come by here in the UK, so I haven't had much experience with them and can't do a direct comparison, but definitely much better than the speakerphones in the Snom units. I would say on a par with the Cisco 7960 (which also graces my desk). ___ -- 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] Music on hold
This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] FXO Cards - T38
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Fernando Berretta wrote: Hi, Could some one let me know if a fax is received through a FXO card connected to * and fax machine is connected to a Linksys FXS device which support T38, is T38 going to be used for faxes which comes from PSTN or go through PSTN ? or because of Asterisk T38 passthrough support it is not possible ? so is for this scenery better to use external FXO gateways with t38 support ? Regards, Fernando ___ -- 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] Music on hold
Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL Voicemail Storage Questions\Errors
I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of voicemail messages into MySQL. It is my understanding that I can only do this via ODBC. I installed per http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to turn on ODBC voicemail storage. Here is the output of some config files: [EMAIL PROTECTED] asterisk]# cat /etc/odbcinst.ini # Example driver definitinions # # # Included in the unixODBC package #[PostgreSQL] #Description= ODBC for PostgreSQL #Driver = /usr/lib/libodbcpsql.so #Setup = /usr/lib/libodbcpsqlS.so #FileUsage = 1 # Driver from the MyODBC package # Setup from the unixODBC package [MySQL] Description = ODBC for MySQL Driver = /usr/lib64/libmyodbc3.so Setup = /usr/lib64/libodbcmyS.so FileUsage = 1 You have new mail in /var/spool/mail/root [EMAIL PROTECTED] asterisk]# cat /etc/odbc.ini [astrealtime] Description = Asterisk realtime FUNC_ODBC access Driver = MySQL Socket = /var/lib/mysql/mysql.sock Server = localhost User= astrealtime Pass= Database= asterisk Option = 3 [EMAIL PROTECTED] asterisk]# cat /etc/asterisk/res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. [asterisk] enabled = no dsn = asterisk ;username = myuser ;password = mypass pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = astrealtime password = pre-connect = yes ; Certain servers, such as MS SQL Server and Sybase use the TDS protocol, which ; limits the number of active queries per connection to 1. By setting up pools ; of connections, Asterisk can be made to work with these servers. [sqlserver] enabled = no dsn = mickeysoft pooling = yes limit = 5 username = oscar password = thegrouch pre-connect = yes [EMAIL PROTECTED] asterisk]# cat /etc/asterisk/voicemail.conf odbcstorage=mysql odbctable=voicemail_messages [EMAIL PROTECTED] asterisk]# asterisk -vvv | grep odbc == Parsing '/etc/asterisk/res_odbc.conf': Found [Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:229 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database [Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:229 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix [Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Feb 22 18:21:46] WARNING[21214]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Feb 22 18:21:46] WARNING[21214]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:302 load_odbc_config: Registered ODBC class 'mysql' dsn-[MySQL-asterisk] [Feb 22 18:21:46] NOTICE[21214]: res_odbc.c:684 load_module: res_odbc loaded. res_odbc.so = (ODBC Resource) [Feb 22 18:21:46] NOTICE[21214]: config.c:1250 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. res_config_odbc.so = (ODBC Configuration) == Parsing '/etc/asterisk/func_odbc.conf': Found func_odbc.so = (ODBC lookups) == Parsing '/etc/asterisk/cdr_odbc.conf': Found cdr_odbc.so = (ODBC CDR Backend) [EMAIL PROTECTED] asterisk]# mysql -u root -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 22 to server version: 5.0.22 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use asterisk; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql describe voicemail_messages; ++-+--+-+-++ | Field | Type| Null | Key | Default | Extra | ++-+--+-+-++ | id | int(11) | NO | PRI | NULL| auto_increment | | msgnum | int(11) | NO | | 0 || | dir| varchar(80) | YES | MUL | NULL|| | context| varchar(80) | YES | | NULL|| | macrocontext | varchar(80) | YES | | NULL|| | callerid | varchar(40) | YES | | NULL|| | origtime | varchar(40) | YES | | NULL|| | duration | varchar(20) | YES | | NULL|| | mailboxuser| varchar(80) | YES | | NULL|
Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors
On Friday 22 February 2008 18:28:56 Mike Hammett wrote: --snip-- [asterisk] enabled = no dsn = asterisk ;username = myuser ;password = mypass pre-connect = yes --snip-- WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! What does enabled mean to you? -- Tilghman ___ -- 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] Music on hold
Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Music on hold
Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Music on hold
it's very odd -I just upgraded to 1.4.18 (from 1.4.17) -removed answer -changed to several other options, still no luck (restarted also) Eric Wieling schreef: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
NOT answering did the trick! Tnx a lot! now it works like it should work! Eric Wieling schreef: Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors
It was my understanding that voicemail.conf referenced MySQL and not asterisk. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 22, 2008 6:56 PM Subject: Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors On Friday 22 February 2008 18:28:56 Mike Hammett wrote: --snip-- [asterisk] enabled = no dsn = asterisk ;username = myuser ;password = mypass pre-connect = yes --snip-- WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! What does enabled mean to you? -- Tilghman ___ -- 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] Music on hold
If not answering fixes the problem then the issue is indications.conf. Try using the indications.conf.sample file included with the Asterisk source code, then stop Asterisk and starting it again. I do not know if indications.conf is reloaded on a reload. Fons van der Beek wrote: NOT answering did the trick! Tnx a lot! now it works like it should work! Eric Wieling schreef: Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] load balancing SIP extensions
I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. The following doc describes using the same extensions across multiple * servers. It requires using realtime, but seems to do what you describe. http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI / Voicemail Que
Thanks Doug, I tried that but it didn't work either... As per Wiki http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it has a statement that starting from 1.4-trunk FLAG must be pass using a pipe sign '|'. I have other Asterisk 1.2 running with FreePBX and I went over the agi code, I saw its passing FLAG using pipe sign '|'. So now I am kinda confused... Cheers, Nitesh Doug Lytle wrote: Nitesh Divecha wrote: ([EMAIL PROTECTED]|b) Any suggestions... By the way I am running Asterisk 1.2.18 I believe under 1.2.x it would be [EMAIL PROTECTED] One of my older dial plans lists: s-BUSY,1,Voicemail([EMAIL PROTECTED]) Doug ___ -- 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] NOKIA E series Phone for SIP-VOIP calling
I have an E61i and it works great with my Asterisk. No extra software needed, everything is built into those phones. Rajeev. ___ -- 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] NOKIA E series Phone for SIP-VOIP calling
Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Thanks, __Yehavi: ___ -- 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] Monitor Asterisk
Hi, I have some experience with Asterisk. What I would like to know is, are there any programmable APIs that we can use to get the information monitored by asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI / Voicemail Que
Nitesh Divecha wrote: Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only Temp Greetings voice mail greetings is played. I am passing the correct parameters for Busy = 'b', Unavailable = 'u' and default goes to Not Answered. I believe the temp greeting will override busy and unavailable greetings. Delete the temp greeting and you should have no problem. Trevor -- Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- 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] FXO Cards - T38
Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- 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] Music on hold
I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? and even with verbose set to 114 (quite big) there are no errormessages indicating that something is wrong with indications (in respect to syntax) Eric Wieling schreef: If not answering fixes the problem then the issue is indications.conf. Try using the indications.conf.sample file included with the Asterisk source code, then stop Asterisk and starting it again. I do not know if indications.conf is reloaded on a reload. Fons van der Beek wrote: NOT answering did the trick! Tnx a lot! now it works like it should work! Eric Wieling schreef: Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Mexico Dids
Hi Robert, DID World Wide has coverage for Saltillo (please see http://www.didww.com/virtual_numbers/Mexico), with flat-rate forwarding to PSTN, VoIP, SIP, H.323, IAX, Skype, MSN or Google Talk. Regards, Gideon From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 22 Feb 2008 18:03:00 -0500 Subject: [asterisk-users] Mexico Dids Hi, I am looking for a did from Saltillo Mexico. Any pointers? robert _ ___ -- 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
