Re: [asterisk-users] Slightly OT: SNOM PoE
I am using the netgear switch 24 ports and 8 ports w/ snom 360 in a 10/100 network w/ no problem. the actual version of firmware of SNOM is 6.5.10 but the phone works w/ previews version. Look in your network. Bruno. Anthony Cennami wrote: Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was curious if anybody has experienced similar issues. Phones will run fine, and then intermittently (and at different times for different ports) the phones will lockup and require a hard reboot. I've read on voip-info that the SNOM phones are apparently sensitive to lower-end network equipment, presumably with PoE only aggravating the problem. Question is, what are people using today to deploy PoE, and more importantly, PoE to SNOM phones? I believe the model we're working with is the SR224P from Linksys, and the entire model line of SNOM (3XX) Could anybody recommend some well-used/tested PoE equipment that you've found successful in your SNOM envionment? Looking for density of 24-ports plus, and ideally some lower end and higher end equipment, to satisfy the needs of the wide variety of customers we do business with? Thanks, anthony -- Anthony Cennami ___ --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 -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --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 Wiki
http://www.voip-info.org/wiki-Asterisk+cmd+Playback you can use google asterisk cmd playback.. bilal ghayyad wrote: Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without alphapatic orgnaization, any advise how to search fast in this website? Regards Bilal Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --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 -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --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 use call transfer
the best way attended transfer. See my feature.conf:
example:
[general]
; Call parking configuration
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in, need to
INCLUDE this in extensions.conf
parkingtime = 45 ; Number of seconds a call can be parked for (default
is 45)
pickupexten = *8
; Max time (ms) between digits for feature activation. Default is 500
featuredigittimeout = 1500
[featuremap]
; Blind transfer, default is pound sign (#)
blindxfer = #
; Attended transfer
atxfer = *7
--END--
Bruno De Luca
Keshav K. wrote:
There is one thing,
just forget that your phone is a snom phone or whatever...
simply to make a blind call transfer press #8, according to the my
feature.conf, default it is #, or you can assign it any, then after
pressing that you will listen a IVR transfer and dial the desired
number followed by the # sign, then you will connect to the new
number, now hangup your phone, and the other two will be connected.
But make sure, that in your extensions.conf you should have the entry
for t, as I have showed in the entry..
Regards,
Keshav
*/satish patel [EMAIL PROTECTED]/* wrote:
but what buttons i will use for call transfer ??? I have SNOM SI
120 phon with transfer button so how to do it ?
*/Keshav K. [EMAIL PROTECTED]/* wrote:
Hi,
To use call tranfer you have to make entry in extension.conf...
exten = _7.,1,Dial(SIP/${EXTEN},20,Ttr)
then in feature.conf
[featuremap]
blindxfer = #8 ; Blind transfer (default is #)
;disconnect = *0 ; Disconnect (default is *)
;automon = *1 ; One Touch Record a.k.a.
Touch Monitor
atxfer = #9; Attended transfer
parkcall = #72; Park call (one step parking)
I'm using this...end its working wonderfully.
--Keshav
*/satish patel [EMAIL PROTECTED]/* wrote:
Dear all
I have beginer in Voip and i have
configured Asterisk server with 100 IP SIP phone ( SNOM )
everything is fine but problem is how to transfer call
from one user to other means i call to some one and then
someone want to transfer call to another person how it is
possible i have also try with feartue.conf but it is now
working i have also read document on voip-info website but
now clear yet can anyone explain me how to asterisk
transfer call from one user to other and what
extention.conf look like is there any change in sip.conf
or extention.conf
Rgd
Satish patel
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Re: [asterisk-users] how to use call transfer
w/ snom u can use the snom transfer and do nothing in asterisk. Or u can use the asterisk transfer (or bind transfer) changing the features.conf (see example) example: [general] ; Call parking configuration parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in, need to INCLUDE this in extensions.conf parkingtime = 45 ; Number of seconds a call can be parked for (default is 45) pickupexten = *8 ; Max time (ms) between digits for feature activation. Default is 500 featuredigittimeout = 1500 [featuremap] ; Blind transfer, default is pound sign (#) blindxfer = # ; Attended transfer atxfer = *7 --END-- Bruno De Luca Gordon Henderson wrote: On Thu, 19 Jul 2007, satish patel wrote: you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call I'd need to read the manual (and I'm sure you're in a better position to do this than I am, as you have the phones and I don't!) You'd normally not need to do anything to the features.conf file to make phone transfers work using the phone features. Gordon Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 18 Jul 2007, satish patel wrote: Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website but now clear yet can anyone explain me how to asterisk transfer call from one user to other and what extention.conf look like is there any change in sip.conf or extention.conf You need to read your phone manual, not the asterisk manual. Every (SIP) phone has it's own ways and means (in addition to the generic features offered by asterisk detailled in features.conf) 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 - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ --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 -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --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 IP 501 is displaying wrong time
Hi, this code is for italian time is inside the sip.cfg file. SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=192.168.0.8 tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=3600 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=1 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ Bruno. Dave Miller wrote: Crazy Boy wrote on 4/19/07 11:41 PM: Thank you for your response. As you said, I set it for -5. But, its displaying wrong time. I don't enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Yeah, there's no way to set the clock except by using an NTP server, so you need to set one. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
I think that the best choice is the snom family... We use all snom in ower office. We tried the Polycom but the support is not so good. Bruno. C F wrote: On 4/13/07, J. Oquendo [EMAIL PROTECTED] wrote: C F wrote: J, Sorry didn't see this email when I wrote the other one (gmail sorts them on a LIFO order). I can agree with you on everything even with the terrible pain of getting Polycoms up and running, but once it is up dont you have less problems with them then with other phones? Isn't the sound quality of the Polycoms better than any of the other phones? Depends on the network sometimes. For clients with anything less than a dedicated-to-VoIP-T1 I would have to disagree. If you do face this situation (someone with low bandwidth), Snom's rock. I did not have that good of an experience with Snoms. I guess I should try again, since it's well over 18 months since I tried last. I stated They aren't the best... but of the whole lot of phones I deal with, they've been thusfar the least problematic. Awesome photo, arn't you having too much fun working? Nah ;) that's like a fraction of junk I play with. At work I have a CC(IE/VP) lab too. 2 3620's 2501, 2522, 3 4500M's, LS1010, Merge ISDN simulator, Pix, Cat 3500's, Netscouts... :D Again I think the Polycom once configure right is quite easy for both the admin and the user. Well, two things come into play so I should have mentioned it. Its best to get a complete picture of what the end user would expect. Once you set those options in XML, unless you're setting up a tftpboot server and can change it, you're hit. I've had far too many instances where clients have ordered them and wanted cosmetic changes that could only be done via the xml files. But what happens when those phones are not booting via tftp. I'm stuck. I either have to have them send me back the phone to make the changes, re-do one and send it back out, or maybe on rare occasions walk someone through having their phone boot via tftp to one my me servers to make those changes. Now ponder this for a minute... Executive John calls me: Can you make this change for me ... I respond Sure can you open up your firewall for me, I will also need you to press x button and enter the following... Even with some so called certified engineers, that becomes cumbersome. This is one point that I have to agree with you, I dread the phone calls that users call me they want just a simple change on a Polycom specific to them. However using FTP, it's only a big deal because of the XML (which also means that I have to document the change, since there is NO way for me to know otherwise that it has a minor change compared to the rest of the users), but it should work nicely remotely as well. All I do before deploying a Polycom phone to a remote site (which is quite easy to walk someone thru it over the phone) is set the FTP Server address, username, and password. Which requires just opening FTP on the server side firewall. That means for security reasons I can't leave it that way, but I could open it up when the user needs a change and have them reboot the phone. Still this is my phone of choice, althoug for the price they should have had much more features when it comes to remapping buttons, or PoE. I must say I have never run into a situation where I had low bandwidth, I always make sure there is at least 768k up, with a less than 150ms latency (not always have been able to meet the later, but never more than 250ms), so can't realy comment on this one. You are pushing me to test that snom again. Will try it. Most of the times if they have their own PBX (I work for a company that does managed PBX's and sells PBX's), and we administrate it, I will set up a squid proxy with only my IP space allowed via ACL's and firewall rules, so I could throw on a proxy on my browser and do it. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simplify
[miprimerejemplo]
exten = _X.,1,Dial(SIP/${exten},30,Ttm)
...
exten = s,1,Dial(SIP/${exten},30,Ttm)
...
Josu Lazkano Lete wrote:
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(2)
exten = 2,103,Hangup
exten = 20100,1,Dial(SIP/20100,30,Ttm)
exten = 20100,2,Hangup
exten = 20100,102,Voicemail(20100)
exten = 20100,103,Hangup
exten = 20200,1,Dial(SIP/20200,30,Ttm)
exten = 20200,2,Hangup
exten = 202000,102,Voicemail(20200)
exten = 20200,103,Hangup
exten = 20300,1,Dial(SIP/20300,30,Ttm)
exten = 20300,2,Hangup
exten = 203000,102,Voicemail(20300)
exten = 20300,103,Hangup
exten = 20400,1,Dial(SIP/20400,30,Ttm)
exten = 20400,2,Hangup
exten = 204000,102,Voicemail(20400)
exten = 20400,103,Hangup
thanks to all
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Re: [asterisk-users] ChanSpy and MeetMe
U can enter to the meetme conference w/ the m option. 'm' --- set monitor only mode (Listen only, no talking) Bruno. [EMAIL PROTECTED] wrote: I have been successful using ChanSpy on a standard Dial call but when attempting to ChanSpy on an incoming call that has been added to a MeetMe conference (attempting to coach an agent that is speaking to a conference of callers) it seems to fail to connect to the channel. Here's the console dump: -- Accepting call from '2154182700' to '3399' on channel 0/18, span 4 -- Executing [EMAIL PROTECTED]:1] Answer(Zap/90-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Read(Zap/90-1, GOTDTMF|demo-instruct|1||1|1) in new stack -- Accepting a maximum of 1 digits. -- Playing 'demo-instruct' (language 'en') -- User entered '5' -- Executing [EMAIL PROTECTED]:3] GotoIf(Zap/90-1, 5?9) in new stack -- Goto (from-internal,3399,9) -- Executing [EMAIL PROTECTED]:9] AGI(Zap/90-1, simpleconf.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/simpleconf.agi -- Playing 'digits/5' (language 'en') -- AGI Script Executing Application: (CHANSPY) Options: (Zap/73|wbq) I verified Zap/73 is the correct channel of the caller currently in the conference I am attempting to ChanSpy on. Has anyone done this before? I apologize in advance if my question lacks the necessary information, I'm new to Asterisk. -George ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ATT Maintenance protocol in PRI connection, no B+D channels available
d-channel is in midle bchan=1-15,17-31 dchan=16 loadzone = it defaultzone = it Kanelbullar wrote: Hi guys, We are experiencing a problem with a T1 PRI connection. After trying a number of variations in the configuration files, the behavior is always the same: no B channels come up and the D channel doesn't appear to be working well. We can see there are ATT Maintenance messages being exchanged by asterisk and the provider, CONNECT and CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring the D and B channels properly up. Are there any messages missing? When we attempt to make a call, we can see the Q.931 SETUP message being sent. But shortly after we are getting a LAPD DISC message, which ends up originating a Q.931 DISCONNECT message, terminating the call. What could be the problem here? * Could there be any configuration issue on our side? * Does libpri provide complete support to the ATT Maintenance protocol or could this connection be incompatible? Any help would be highly appreciated. Many thanks in advance, Paulo PS: Configuration files, messages and pri debug snippets follow zaptel.conf loadzone = us defaultzone=us #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 PRI_T1 span=1,0,0,esf,b8zs,crc4 bchan=1-23 dchan=24 zapata.conf [channels] group = 0 usecallingpres = yes switchtype = national context = inbound signalling = pri_cpe usecallerid = yes channel = 1-23 messages -- Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will be lost. Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get merged Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice [...] pri debug span -- [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 005 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT (7) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 005 0: 0 N(R): 006 P: 0 10 bytes of data -- ACKing all packets from 5 to (but not including) 6 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: ATT Maintenance (3) len=10 Call Ref: len= 1 (reference 0/0x0) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [01 01 c0] IE: Change Status (len = 3) [18 01 ac] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 1 ChanSel: As indicated in following octets ] (...) Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 06 21 80 37 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '7100' ] [70 0b a1 35 38 35 34 31 39 37 39 39 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5854197995' ] (...) [ 02 01 53 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 2 P/F: 1 M2: 0 11: 3 [ DISC (disconnect) ] 0 bytes of data -- Got Disconnect from peer.
Re: [Asterisk-Users] Dial out .call files File permissions??
do u need to give the permission for user asterisk to uour file. Bruno. Marco Mouta wrote: Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] register = 2345:[EMAIL PROTECTED] doesn't care about port
file sip.conf:
register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
example:
register = 531:[EMAIL PROTECTED]:5061/1234
file extensions.conf
exten = extension,1,1,Dial(number)
exten = extension,1,2,HangUp
example:
exten = 1234,1,1,Dial(SIP/1)
exten = 1234,1,2,HangUp
to call
file sip.conf:
[**NAME**]
type=peer
secret=**PSW**
username=**USER**
port=**PORT**
host=**HOST**
fromuser=**USER**
fromdomain=**DOMAIN**
nat=yes
* ESEMPIO:
[messagenet-out]
type=peer
secret=pwd
username=nmb
port=5061
host=sip.messagenet.it
fromuser=nmb
fromdomain=sip.messagenet.it
nat=yes
file extensions.conf
exten = extension,1,Dial(number,30,r)
* ESEMPIO:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
[EMAIL PROTECTED] wrote:
Hi,
to register my Asterisk with a SIP provider I use the following
syntax, as shown in the default sip.conf:
register = 2345:[EMAIL PROTECTED]
where
[sip_proxy]
type=peer
context=from-messagenet
host=sip.messagenet.it
port=5061 - please note this one!!!
5061 is provider's port I have to register to.
This also would work for me:
register = 2345:[EMAIL PROTECTED]:5061
but I need the other syntax 'cause I *have* to specify a different
context for incoming calls rather than the default one in sip.conf.
Well, sip show registry shows:
Host Username Refresh State
sip_proxy:5060 *** 105 Registered
As you can see, Asterisk didn't care about port value 5061.
However, sip show peer sip_proxy shows:
ToHost : sip.messagenet.it
Addr-IP : 212.97.59.76 Port 5061
-
What is wrong, please?
Should I report this behaviour as a bug? Maybe a feature request?
Cheers,
Alex
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Re: [Asterisk-Users] dell and digium hardware
The very problem is that DELL in the small one, block the IRQ. And this can make conflict to the cards. Bruno. Klaus Darilion wrote: Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking por a provider to work with asterisk
http://freevoip.gedameurope.com Jason Brashear wrote: I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] firmware update polycom 500 / dial problem
U need to set your digitmap. Morel Mosolff wrote: Hi, sorry - I know that problem is not directly related to asterisk but mabe someone can help anyway. After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is mostly not possible to dial numbers with leading zeros like 0018... If you do so you see on the diplay an number like that: 1800 an the cursor is on the first position. But if you dial the number (press the buttons) without lifting the handset everything is ok...strange Thank you for any help, morel -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX test service
Try FWD. Gabor Horvath wrote: Dear Asterisk users, can you suggest me a free service where I can test my IAX trunks? Thank you. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-users] VoiceMail help
did u set the mailserver? Bruno. Fabio Montemaggiore wrote: I don't receveid e-mail with voicemail. When I dial 2 with telephone, Asterisk record message but don't send a e-mail at the mailbox. Why? I have configuration this file. In the voicemail.conf [general] attach=yes format=wav skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 sendvoicemail=yes [zonemessages] italia=Europe/Rome|'vm-received' Q 'digit/at' HMP [101] 100 = 100,100,[EMAIL PROTECTED],,|attach=yes In the dialplan: exten = 2,1,Answer exten = 2,2,Wait(1) exten = 2,3,VoiceMail(u100) exten = 2,4,Playback(vm-goodbye) exten = 2,5,Hangup ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom software
I will send them to u. Bruno De Luca Bartosz Jozwiak wrote: Dear users, It might be slightly off topic. I own couple 500 and 600 Polycom SoundPoint IP phones and need to download new software for them. The phones has been purchased from voipsupply.com Is there a way to download such a software without becoming certified reseller? Thanks, Bartosz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
this can help u: SIP.CONF [1] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 1 pickupgroup = 1 callerid = "Bruno De Luca 1" 1 secret = 1234 mailbox = 1 context=1 [2] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 2 pickupgroup = 2 callerid = "Bruno De Luca 2" 2 secret = 1234 mailbox = 2 context=2 [3] ... context=1 [4] ... context=2 EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ________ BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] add 0 (zero) to incoming callerID - how?
Use
${CHANNEL} to get the number
then
SetCIDNum() to set the new number w/ zero.
Tomasz Chmielewski wrote:
I have an
asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial feature in the telephone (because the phone doesn't
know
that it should add 0 before the number).
So the idea is to manipulate the incoming callerID number, and to add a
0 before it.
This way the telephone user will be able to callback/redial.
How can I manipulate the incoming callerID number (and add 0 before
it)?
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] add 0 (zero) to incoming callerID - how?
Correct: Use ${DNID} to get the number. I'm sorry.
Bruno.
Bruno De Luca wrote:
Use
${CHANNEL} to get the number
then
SetCIDNum() to set the new number w/ zero.
Tomasz Chmielewski wrote:
I have an
asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial feature in the telephone (because the phone doesn't
know
that it should add 0 before the number).
So the idea is to manipulate the incoming callerID number, and to add a
0 before it.
This way the telephone user will be able to callback/redial.
How can I manipulate the incoming callerID number (and add 0 before
it)?
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] GotoIf sample...
my example:
[i_day]
exten = _X.,1,Answer
exten = _X.,2,NoOp(${DNID})
; FIMLA_TEL
exten = _X.,3,GotoIf($[${DNID} = ${TEL_FIMLA_1}]?4:8) ; Se tel_fimla_1 = call FIMLA
exten = _X.,4,Dial(${FIMLA},${RING_4X_TIME})
exten = _X.,5,GotoIf($[${DIALSTATUS} = NOANSWER]?6:15)
exten = _X.,6,SetCIDName(FIMLA)
exten = _X.,7,Dial(${SOCON},${RING_TIME})
; FIMLA_FAX
;exten = _X.,7,GotoIf($[${DNID} = ${TEL_FIMLA_2}]?8:9) ; Se tel_fimla_2 = call FAX_FIMLA
;exten = _X.,8,Dial(${FAX_FIMLA},${RING_TIME})
; SOCON_TEL
exten = _X.,8,GotoIf($[${DNID} = ${TEL_SOCON_1}] ? 11:9)
exten = _X.,9,GotoIf($[${DNID} = ${TEL_SOCON_2}] ? 11:10)
exten = _X.,10,GotoIf($[${DNID} = ${TEL_SOCON_3}] ? 11:15)
exten = _X.,11,Dial(${SOCON},${RING_4X_TIME})
exten = _X.,12,GotoIf($[${DIALSTATUS} = NOANSWER]?13:15)
exten = _X.,13,SetCIDName(SOCON)
exten = _X.,14,Dial(${FIMLA},${RING_TIME})
; FIMLA_FAX
; TODO: IMPLEMENTARE.
exten = _X.,15,Hangup
exten = t,1,Hangup
Terry Wilson wrote:
On 9/5/05, ryan nalupa
[EMAIL PROTECTED]
wrote:
hi everyone. can anyone provide me concrete examples on how to
use
the GotoIf application? can't figure out how to use it in my dialplan
coz im having errorsthanks! : )
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoIf
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BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] Receiving Calls from FWD Network using IAX2
in iax.conf devi anche mettere questa riga per ogni fwd: register = FWDNumber:[EMAIL PROTECTED] Bruno. kswail wrote: Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a Call Disconnected 486 error. What's odd is that I don't see any debug info from the console (iax2 debug). I've tried forwarding UDP port 4569 to my Asterisk box and no diff. Anyone have any advice? Cheers! kswail === Here are relevant parts of my configs --- iax.conf --- register=x:[EMAIL PROTECTED] [fwd] username=x type=peer secret= qualify=yes host=iax2.fwdnet.net auth=md5 [fwd-in] type=user inkeys=freeworlddialup context=from-pstn auth=rsa === Here is output from the asterisk console as it pertains to IAX2 --- asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569x 00.00.00.244:4569 60 Registered --- asterisk*CLI iax2 show peers Name/UsernameHost Mask Port Status fwd/x65.39.205.121 (S) 255.255.255.255 4569 OK (15 ms) --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] same extension on multiple sip phones?
U can use this way in extensions.conf:
exten = 2,1,Dial(${BRUNO_FGA}${GIORGIO_FGA},${RING_TIME}) ; supp-tecnico
Bruno
Kevin Hanson wrote:
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
A client wants an incoming call to ring at the receptionists desk and
also at their desk. If the receptionist is in it will be answered
there and put on hold followed by a Joe, you have a call on line 1.
Is there a way to do this w/ asterisk? I've played with two phones
with same sip registration and it seems the last one to register is
the one asterisk recognizes.
Thanks,
Kevin
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BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] invalid extension dilemma
u can use this:
exten = i,1,Playback(invalid_selection)
exten = i,2,Goto(inbound_menu,_X.,1)
Bruno.
Joseph wrote:
Ho do you folks solve the problem with invalid extension when someone
dials a wrong number?
For example if somebody dial prefix _7 I want to allow tall free
numbers from that line but not a long distance. However, if somebody
dial
wrong number I want to play invalid extension instead of congestion.
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
[goto-dialout]
exten = _9.,1,SetMusicOnHold(loud)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9.,3,Hangup()
exten = _71800XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71866XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71877XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71888XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,1,SetMusicOnHold(loud)
exten = _7NXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,3,Hangup()
include = invalid
[invalid]
exten = _.,1,NoCDR()
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
exten = 1000,4,Hangup()
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
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Re: [Asterisk-Users] Polycom Soundpoint 500
Try to control the file in the server... i have seen that this phone change the server file in an wrong way... Bruno. Brent Davidson wrote: I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug 2 15:48:49 NOTICE[11606]: chan_sip.c:9405 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for 'XX.XX.XX.XX' Aug 2 15:48:50 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=CD6D3F82-1211688D for SUBSCRIBE Aug 2 15:48:52 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=CFBF905B-DD972A1A for SUBSCRIBE Aug 2 15:48:53 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=24939F70-451E5F93 for SUBSCRIBE Aug 2 15:48:55 NOTICE[11606]: chan_sip.c:9405 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for 'XX.XX.XX.XX' Aug 2 15:48:56 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=2E59724E-73F0A849 for SUBSCRIBE The phone has two lines, extension 7202 and 7203. I don't receive any messages regarding 7203, and the two sip profiles are identical in the sip.conf file (with teh exception of substituting 7202 for for 7203) and I have retyped the password into the phone more times than I can count. Now the odd thing is that the phone can make and receive calls, they are just very choppy when calling IAX extensions. When the calls go to/from the Polycom from/to a Zap channel, the calls are perfectly clear. I am completely lost at this point. Any ideas? Thanks, Brent Davidson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange dial problem with polycom 501
try to see if u have set at sip.conf *dtmfmode=rfc2833* Michael George wrote: I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above, if just typed into the phone, might end up: 9507291234. Other times the cursor might jump right back to the beginning of the number. This doesn't happen when they enter the number and the press dial, so it seems to be a digitmap problem. However, the digitmap is nearly the same as what I've used on IP-500s in the past. It is: [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T [Actually it was [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T -- I don't know where that space came from, but I'll take it out and test again today.] Are there any obvious problems with that digitmap? Anything else that I should take a look at? Thank you. -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...
U are using SIP ? if yes set *type=friend*
Bruno.
Mauro Zanin wrote:
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off
and now it executes:
*exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})*
when it runs, the mail box number is asked and password too. I
expected no question were made, because I inserted CALLERIDNUMBER and
s in front of box number.
Anybody knows why?
Thank to you all, very kind members of this list!
Ciao
Mauro
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FGA Software
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Re: [Asterisk-Users] Attended transfer not working (atxfer)
When u trasfer u need: Trasfer key sequence trasferee numer talk trasfer key sequence Bruno. Damian Minkov wrote: While on conversation with another party, I dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep' (language 'en') And then in the console of asterisk is wrote : -- Executing Hangup(Transfered/SIP/8008-432aZOMBIE, ) in new stack Number SIP/8008 is the first originator of the call which must be connected to the transferee. Any ideas? I use CVS of asterisk from 2005-06-16 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exten = fax in [macro-blah]
Try this:
exten = fax,1,Dial(${FAX})
exten = fax,2,Congestion
exten = fax,102,Congestion
Bruno.
Eric Wieling aka ManxPower wrote:
It seems that exten = fax does not work in a macro.
Asterisk detects the fax, since it complains about no fax extension,
but I have an exten = fax in the macro.
Has anyone else experienced this?
--Eric
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FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
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Re: [Asterisk-Users] Polycom IP600 - Flashing clock and date?
U can set your linux to do this work. An SNTP Server. Bruno. Billy Dunn wrote: [EMAIL PROTECTED] wrote: There should be a NTP setting. Setup Network Time Protocol. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This was a pain in the butt for me. In fact, I only was able to get it going by pointing the SNTP server to pool.ntp.org and making sure the DNS entries were correct. That works, but it's not a great solution. When the phone is flashing, that means it cannot contact the SNTP servers. Ideally it should talk to a local NTP server on your network, but I have yet to see that work (but I'm only two weeks into Asterisk too). Good luck. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft Phone
U can try the firefly. this softphone can be used w/ more then 1 line... https://www.virbiage.com/download.php To configurate this in asterisk is like a normal sip phone or iax phone... -SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) videosupport=yes ; Support Video disallow=all ; Disable all codecs allow=alaw allow=ulaw allow=g726 allow=g723.1 allow=g729 allow=gsm context = sip ; Send unknown SIP callers to this context ; - SIP ACOUNT - [INT_10] type=friend qualify=no ;500 host=dynamic dtmfmode=inband callgroup=2 pickupgroup=2 callerid=Gianni 10 secret=segredo BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uk Caller id
this is an italian code and works... try it. [channels] ; -- canale 4 -- language=en faxdetect=both musiconhold=default group=2 canpark=yes context=inbound signalling=fxs_ks usecallerid=no ; echo cancel echocancel=128 ; range from 32 to 256(=echo 100%) echocancelwhenbridged=yes ; yes = 400 msec echotraining=200 channel=4 Bruno De Luca Graziosi Chris Thompson wrote: Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen 0 (-20) Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: Success Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=uk callerid=asreceived cidsignalling=v23 cidstart=usehist callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no progzone=uk musiconhold=default ; incoming channels signalling=fxs_ks group=2 context=incoming channel = 1-2 ; outgoing channels signalling=fxo_ks group=1 context=outgoing channel = 3 Thanks loads Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
I know. but u can't disable the USB controller always. If u have an server w/ others functions... Bruno De Luca Graziosi DELL computers ussualy has got IRQ conflicts with the USB and slots PCI. If you disable the USB controller from BIOS you get a perfect server. I have tried several PowerEdge 2850 like Asterisk dedicated server and it's running perfectly. I have tried IBM xServer 226 and 346 and the IRQ conflicts (network with slots PCI and with video card) make noises, echos and cuts off . :( Elio Rojano == Avanzada7 -VoIP Departure- http://www.avanzada7.com/ We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
