[asterisk-users] Which PCIe cards work with North American BRI & Asterisk?
Which PCIe cards work with North American BRI & Asterisk? Digium & Sangoma don't support it, according to everything I've read (their manuals and their tech support). I think I need National, maybe National 1 or National 2. I already got an NT-1. I'm getting a few ISDN phones to test the circuit. I ordered a Dialogic Diva Diva 4BRI - 8 PCI-E 4 Ports Quad 803-031-02B Gu... but I'm concerned that it does not work out of the box with Asterisk and requires proprietary closed firmware or software. Brad Allen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.5.0
This is what worked for me on a CentOS 6.4 32-bit system. libuuid-2.17.2-12.9.el6_4.3.i686 libuuid-devel-2.17.2-12.9.el6_4.3.i686 - this is the one you need uuid-1.6.1-10.el6.i686 uuid-devel-1.6.1-10.el6.i686 You will want the x86_64 versions of these for your system... On Mon, Aug 12, 2013 at 1:37 PM, Doug Lytle supp...@drdos.info wrote: I did as you suggested, the make menuselect showed XXX by res_rtp_asterisk and said depends on uuid(E). On my Mageia system: rpm -qa|grep -i uuid libuuid-devel-2.17.1-5.2mdv2010.2 libuuid1-2.17.1-5.2mdv2010.2 So maybe you also have a libuuid-devel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Light-weight voice recognition for IVR
Hello list, 'Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like: 0 - 9 yes no (maybe * and # for some people) The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like: For Archie, press or say 1 now For Veronica, press or say 2 now For Jughead, press or say 3 now (etc.) You have selected option 2 for Veronica, press 1 or say yes if this is correct. If a voice response was received (not a DTMF key press) indeterminate, some status would be useful (beyond just a timeout). It would be great if this was simple to code into the dialplan, much like like the current background/wait model for keypresses. Low cost or free would be nice too! Thanks for any suggestions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 dtmf not recognised
On Sat, May 25, 2013 at 10:32 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Finally got it working with 3 attempts by the fialplan,
exten = 300,1,Playback(letters/a)
exten = 300,n,Set(gottries=0)
exten = 300,n(getmore),Set(rightPIN=1)
exten = 300,n,Read(inPIN,,1,skip,3,3) ; Attempts for 3 times with 3
seconds of timeout
exten = 300,n(gotdigit),GotoIf($[${inPIN} =
${rightPIN}]?pin-accepted,1)
exten = 300,n,Set(gottries=$[${gottries}+1];
exten = 300,n,GotoIf($[${LEN(${inPIN})} == 0]?reallynothing:gotdigit)
exten = 300,n(reallynothing),GotoIf($[${gottries}3]?done:getmore) ;
Attempts for 3 tries if greater than 3 then it will come out or else
getmore will called
exten = 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so
play badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
Thanks
On 25 May 2013 15:38, Gopalakrishnan N gopalakrishnan...@gmail.com
wrote:
Am using Read application to get the digit, since its recognizing... I
would like to get for 3 attempts and then after 3rd attempt it has to
playback some different message like entries exceeded.
My dialplan as,
exten = 100,1(begin),Playback(letters/a)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3
seconds of timeout
exten = 100,n,GotoIf($[${inPIN} = ${rightPIN}]?pin-accepted,1)
exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
what happens its keep on asking to enter digit If my DTMF didnt match. Do
i need to use any return function... ?
Actually my goal is to ask for 3 times and if not matched then return to
some other application.
Thanks in advance.
On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
With Asterisk 1.8 I got it working.
Regards
On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:
dtmfmode=auto
dtmfmode=info
or
dtmfmode=rfc2833
Doug
--
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Temporary Safety, deserve neither Liberty nor Safety.
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Syntax check:
exten = 300,n,Set(gottries=$[${gottries}+1];
should be:
exten = 300,n,Set(gottries=$[${gottries}+1])
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[asterisk-users] Failed to authenticate device Ext 110
I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows Forbidden, and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-0004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device Ext 110 sip:110@192.168.6.2;tag=1130259112 Asterisk 192.168.6.2 OpenVPN on router 10.8.0.1 Remote Yealink phone 10.8.0.6 The remote phone shows as being registered: PBX*CLI sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN Also, if there is voicemail in the mailbox for 110, the phone's message light is lit and it beeps periodically. toshi*CLI sip show peer 110 * Name : 110 Description : Yealink OpenVPN Secret : Set MD5Secret: Not set Remote Secret: Not set Context : remote-phones Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : 110 VM Extension : asterisk LastMsgsSent : 1/0 Call limit : 4 Max forwards : 0 Dynamic : Yes Callerid : Ext 110 110 MaxCallBR: 384 kbps Expire : 608 Insecure : no Force rport : No Symmetric RTP: No ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: Yes Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.8.0.6:5062 Defaddr-IP : 10.8.0.6:5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 110 SIP Options : (none) Codecs : (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : OK (237 ms) Useragent: Yealink SIP-T28P 2.61.23.3 00:15:65:xx.xx.xx Reg. Contact : sip:110@10.8.0.6:5062 Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No sip.conf: [110] context=remote-phones type=peer host=dynamic qualify=1500 canreinvite=no dtmfmode=rfc2833 progressinband=no callgroup=1 pickupgroup=1 ; We can do call pickup for call group 1 call-limit=4 busy-level=1 qualify=yes deny=0.0.0.0/0.0.0.0 permit=0.0.0.0/0.0.0.0 nat=no qualify=8000 description=Yealink OpenVPN defaultuser=110 secret=x callerid=Ext 110 110 mailbox=110 defaultip=10.8.0.6 port=5060 disallow=all allow=ulaw Any suggestions on what might be happening here, and how it could be resolved? THANKS ALL! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate device Ext 110
On Tue, May 21, 2013 at 11:26 AM, Matthew J. Roth mr...@imminc.com wrote: asterisk users wrote: I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows Forbidden, and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-0004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device Ext 110 sip:110@192.168.6.2 ;tag=1130259112 Asterisk 192.168.6.2 OpenVPN on router 10.8.0.1 Remote Yealink phone 10.8.0.6 The remote phone shows as being registered: PBX*CLI sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN Also, if there is voicemail in the mailbox for 110, the phone's message light is lit and it beeps periodically. ... Any suggestions on what might be happening here, and how it could be resolved? That is quite strange. Please provide SIP traces of the dialogs between Asterisk and the phone in the following two scenarios: 1) Phone registering to Asterisk (presumably successful) 2) Phone dialing to Asterisk (presumably unsuccessful) Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- Registration trace (note that extension 88 is the voicemail extension, which the phone registers to also for MWI) -- http://pastebin.com/c3H700wa Call trace: |Time | 10.8.0.6 | | | | 192.168.6.2 | |268.693661| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP From: Ext 110 sip:110@192.168.6.2 To:sip:88@192.168.6.2 | |(1024) -- (5060) | |268.694449| 401 Unauthorized |SIP Status | |(1024) -- (5060) | |268.914195| ACK | |SIP Request | |(1024) -- (5060) | |268.945115| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP From: Ext 110 sip:110@192.168.6.2 To:sip:88@192.168.6.2 | |(1024) -- (5060) | |268.945717| 403 Forbidden |SIP Status | |(1024) -- (5060) | |269.041417| ACK | |SIP Request | |(1024) -- (5060) | I'm also confused by the reference in sip show peers to port 5062, as I can't see that anywhere in the configuration of either the phone or in sip.conf. All the other phones show port 5060 in the sip show peers output. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk voicemail minimum length / silence settings
What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ; How many seconds of silence before we end the recording maxsilence=10 With these settings, I'm getting the following warning message. WARNING[21671] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages What are the right settings for this situation? Thanks all! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help designing implementation
Hi, I'd like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions. I want to setup two Asterisk servers that are linked to each other: - The first server would be my external (public) server and would live in a real data center. The second server would be my internal (private) server and would live in my house. - The external server would receive all incoming calls and handle the voice mail stuff. - The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server. I also want to add the following additional functionality: - If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can't reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system. - If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call. - I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM). I would like specify in a white list specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents). - I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages. Is all of this possible? If not, which part's are not (and how much work do you think would be needed to make those parts work)? -- Thanks, Dyweni -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium IP Phones - Teleworker Capability?
We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Thanks for all comments! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/14/2012 04:57 PM, asterisk users wrote: We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? Yes, I believe they do :-) The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. Digium phones can do something similar, and in an upcoming firmware release, there will even be features available to make this happen on a fairly automatic basis. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. Digium phones speak SIP and RTP to the server, just like pretty much any other SIP phone. They employ many modern NAT traversal techniques and should work in most network situations. They don't currently provide encryption for signaling and media, though. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. I don't understand this; SIP phones don't require this at all. The phone is an intelligent device on its own. If there is no network connectivity to the server, then calls cannot be placed or received, but once connectivity is restored, operation would be back to normal. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). Yes, this is supported. These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- This is pretty good news, overall. To comment on Kevin's points: - The end-to-end encryption is important to us, because client-ID-sensitive information is part of our environment. Something like built-in OpenVPN would work for us, if that were an option. - Being fault-tolerant (of less than perfect DSL and rural-wireless connections - if the boss is at his cabin, for instance) and being very user-friendly about it is really important to end users. Minet has a heart-beat mechanism so that if the connection goes down between the phone and the switch, the display shows it. Of course, calls get diverted to voicemail during that period. If something is not working in the network, the user is informed about it, and when it is fixed, everything continues, including button DSS status updates, voicemail WMI, etc. On typical SIP phones, everything looks normal until you go to use it, then there is no dialtone, or you just get dead-air on the handset). Our users are pretty demanding, and want a utility-grade solution that will always work - for them. - Most of it, I think. Give them a try! Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] downloads.asterisk.org appears to be down right now
Connecting to downloads.asterisk.org 76.164.171.233|:80... failed: Connection timed out. Just fyi. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting outbound PRI Callerid with Asterisk 10.0.0-beta2
Hello all, I'm having trouble setting the callerid name and number independently with the following configuration: Asterisk 10.0.0-beta2 DAHDI Version: 2.5.0 Echo Canceller: HWEC, MG2 libpri version: 1.4.12 Allstream PRI 23+D / dms100 Test cases: (1) Using the form: same = n,Set(CALLERID(all)=ABCD COMPANY 519111) same = n,Dial(Dahdi/G1/519333) both the caller ID and name are passed correctly to the called party (2) However, if we try to set the number and name separately same = n,Set(CALLERID(name)=ABCD COMPANY) same = n,Set(CALLERID(number)=519111) same = n,Dial(Dahdi/G1/519333) then with some called numbers, we get a congestion message (All circuits are busy) unless the first CALLERID(name) line is commented out. With other numbers the call completes, but the received callerid shows as: CALLERID(all): \ 519111 Is there some subtle difference between these two methods, or would anyone have some experience with this? Thanks for any comments, and Happy Friday! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outbound PRI Callerid with Asterisk 10.0.0-beta2
Same result: Executing [s@incoming:3] NoOp(SIP/choicetel-0092, CALLERID(all): \ 519111) in new stack On Fri, Nov 18, 2011 at 2:05 PM, Danny Nicholas da...@debsinc.com wrote: Just a hunch same = n,Set(CALLERID(name)=AB\CD COMPANY) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users Sent: Friday, November 18, 2011 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Setting outbound PRI Callerid with Asterisk 10.0.0-beta2 Hello all, I'm having trouble setting the callerid name and number independently with the following configuration: Asterisk 10.0.0-beta2 DAHDI Version: 2.5.0 Echo Canceller: HWEC, MG2 libpri version: 1.4.12 Allstream PRI 23+D / dms100 Test cases: (1) Using the form: same = n,Set(CALLERID(all)=ABCD COMPANY 519111) same = n,Dial(Dahdi/G1/519333) both the caller ID and name are passed correctly to the called party (2) However, if we try to set the number and name separately same = n,Set(CALLERID(name)=ABCD COMPANY) same = n,Set(CALLERID(number)=519111) same = n,Dial(Dahdi/G1/519333) then with some called numbers, we get a congestion message (All circuits are busy) unless the first CALLERID(name) line is commented out. With other numbers the call completes, but the received callerid shows as: CALLERID(all): \ 519111 Is there some subtle difference between these two methods, or would anyone have some experience with this? Thanks for any comments, and Happy Friday! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outbound PRI Callerid with Asterisk 10.0.0-beta2
Thank you. Omitting the quotes on CALLERID(NAME) line seems to be the solution. This works: same = n,Set(CALLERID(name)=ABCD COMPANY) same = n,Set(CALLERID(number)=519111) same = n,Dial(Dahdi/G1/519333) It is strange, though, that with quotes in, the call actually fails with congestion, and only with certain numbers dialed (all 10-digit local area dialing). Some sort of error message would be nice. :-) This help is much appreciated! On Fri, Nov 18, 2011 at 2:43 PM, Richard Mudgett rmudg...@digium.com wrote: Hello all, I'm having trouble setting the callerid name and number independently with the following configuration: Asterisk 10.0.0-beta2 DAHDI Version: 2.5.0 Echo Canceller: HWEC, MG2 libpri version: 1.4.12 Allstream PRI 23+D / dms100 Test cases: (1) Using the form: same = n,Set(CALLERID(all)=ABCD COMPANY 519111) same = n,Dial(Dahdi/G1/519333) both the caller ID and name are passed correctly to the called party (2) However, if we try to set the number and name separately same = n,Set(CALLERID(name)=ABCD COMPANY) You do not need to quote the name here. Otherwise, the quotes are included as part of the name. Everything between the '=' and closing ')' less leading and trailing spaces is part of the name. For SIP this may be detrimental to the message format unless the quotes get escaped. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663)
exten = 2663,1,Answer
same = n,Wait(1)
same = n,Authenticate(143382)
;Record conference callscount: ${count} --)
same = n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which
Asterisk version are you using?
*CLI core show function CONFBRIDGE_INFO
No function by that name registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didn’t want to duplicate the meetme_count
function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins, so
in the example below, for example, how can I get the current user count in
ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663)
exten = 2663,1,Answer
same = n,Wait(1)
same = n,Authenticate(143382)
;Record conference callscount: ${count} --)
same = n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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Re: [asterisk-users] ConfBridge 1.6.20 user count
confbridge(xxx,c) is a blocking call, so you can't get status back
until that command completes. Time to upgrade to 10.0.beta2 I
guess...
On Wed, Nov 9, 2011 at 12:47 PM, Danny Nicholas da...@debsinc.com wrote:
10.0.beta2. Have you tried confbridge(xxx,c)? This joins and announces
count, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
Unfortunately, that function doesn't seem to be in 1.6.20, which Asterisk
version are you using?
*CLI core show function CONFBRIDGE_INFO No function by that name
registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked') and for invalid conference identifiers.
[Syntax]
CONFBRIDGE_INFO(type,conf)
[Arguments]
type
Type can be 'parties', 'admins', 'marked', or 'locked'.
conf
Conf refers to the name of the conference being referenced.
Guess the developers of confbridge didn’t want to duplicate the
meetme_count function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi all,
I'm using ConfBridge within Asterisk 1.6.20 and want to record the
conference, so I'd like to start the recording when the second user joins,
so in the example below, for example, how can I get the current user count
in ConfBridge 3000?
[conferences]
;authenticated conference (ext C-O-N-F = 2663) exten = 2663,1,Answer
same = n,Wait(1) same = n,Authenticate(143382)
;Record conference callscount: ${count} --) same =
n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
same = n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
same =
n,ExecIf($[${count}=2]?Monitor(wav,conf-${CALLERID(num)}-${DATETIME},bm))
-- count?
same = n(conf),ConfBridge(3000,Ms)
same = n,Playback(goodbye)
same = n,Hangup
Thanks for any ideas!
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[asterisk-users] MixMonitor not recording in version 1.8
Greetings.
Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work
ok. Except for one thing.
I have a call to MixMonitor. This is implementing a dictaphone kind of app.
With forwarding recordings to email and storing them on the server.
The process works so that we dial into Asterisk and answer the phone,
initiate MixMontior and WaitExten until recording finishes.
Problem is that in 1.8 the MixMonitor does not begin recording, ever (when
applied as shown below). I've tried MixMonitor on the same server with
bridged channels and this is no problem and works as expected.
Question is, is there a way to force MixMonitor to work on 1.8 as it used to
on 1.4.22?
Dial plan (AEL) is as follows (excerpts):
// BEGIN OF SAMPLE
incoming {
555 = {
jump 0...@dicta;
}
}
dicta {
= {
Answer(1000); // Slight initial pause to allow audio to
balance
Playback(beep);
// ***
// THIS IS WHERE the problem lies.
// This call does NOT start recording at this time!
// It used to work, in 1.4.22. But in 1.8 it does not.
// ***
MixMonitor(myfilename.alaw,,mv myfilename.alaw
myfinishedfilename.alaw);
jump 0...@dicta-while-recording;
}
}
dicta-while-recording {
0001 = {
WaitExten(400); // This is effectively the maximum length of
a recording!
}
}
// END OF SAMPLE
Any help is greatly appreciated.
Best regards,
Baldvin
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Re: [asterisk-users] iax calls via checkbox.cc
Joseph wrote: Is anybody using checkbox.cc to make iax2 calls? They have recently did some changes and my calls no no longer go through. They don't have a best service either, not replying to emails.. I don't know about that company, but since you are sounding unhappy with them, have you looked into callwith.us? -Brandon Broyles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax calls via checkbox.cc
Joseph wrote: On 05/11/10 18:31, John Novack wrote: lists-asterisk-us...@yoinks.net wrote: Joseph wrote: Is anybody using checkbox.cc to make iax2 calls? They have recently did some changes and my calls no no longer go through. They don't have a best service either, not replying to emails.. I don't know about that company, but since you are sounding unhappy with them, have you looked into callwith.us? -Brandon Broyles No IAX No number porting Otherwise the web site talks a good game John Novack No IAX I remember now; I used to have an account with them and the call quality was OK but they discontinued the IAX. An account on callwith.us can be funded via Paypal or credit card. There may be other methods also. Yeah, they don't IAX. And they are only a SIP termination provider. For SIP origination I use ipcomms.net. -Brandon Broyles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
Thank you all for your input into this question. It is very helpful to get your opinion and experience with this matter. I mean in my case a single server application. And what I'm probably going to have to do is use AMI via either a windows .net application that will parse and monitor the ami stream or (which I'd much rather prefer, but lack experience and knowledge to complete) some Linux based implementation that could turn an AMI event into a HTTP request formatted to my requirements in real-time. I'm now thinking about looking into python or perl or something to try and get this going. tnx! Baldvin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 6. júlí 2009 12:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the best way to share extension state 2009/7/6 asterisk-us...@rogg.is Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What do you exactly mean by applications ? Do you mean a single server application or several instances of client applications ? ... Sincerely, Baldvin attachment: winmail.dat___ -- 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] What is the best way to share extension state
Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What I'm after is the best way to have Asterisk update a central repository with the state of each extension configured in the local Asterisk setup. To try and explain what I am trying to achieve, Imagine for example if asterisk would call a url like this: http://myserver/updatestatus.php?ext=101state=idle; http://myserver/updatestatus.php?ext=101state=ringing; http://myserver/updatestatus.php?ext=101state=occupied; for every state change of every extensions. I've already looked closely at a few ways to do this. The closest I've come so far is thinking in this direction: 1) Set up a process that does SIP SUBSCRIBE to read hints from Asterisk extensions. 2) Use jabber integration to somehow achieve this. 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives me what I want. Possible problems with the things I've considered so far may be for example: o Extensions that are part of a queue (making sure state is reflected even if the ext is ringing as part of a queue ringing). o Making sure the process can recover even if the monitoring entity needs to be restarted. I'm very curious to hear what your take on this is and if this has perhaps been solved elegantly already? Thank you for considering this question and your time spent thinking about this and possibly replying with your thoughts. Sincerely, Baldvin ___ -- 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] Determining picked up line from multiple line ring
Hi all, I've looked at the various variables but can't seem to find a way to determine which line was picked up in a multi-line ring. For example, in this excerpt from my asterisk logging: -- Executing [5558280...@inbound:52] Dial(SIP/proxy3-05ac9180, SIP/1555...@proxy1SIP/1555...@proxy1|18|r) in new stack -- Called 1555...@proxy1 -- Called 1555...@proxy1 -- SIP/proxy1-05af5ca0 is making progress passing it to SIP/proxy3-05ac9180 -- SIP/proxy1-05acaae0 is making progress passing it to SIP/proxy3-05ac9180 -- SIP/proxy1-05acaae0 answered SIP/proxy3-05ac9180 -- Packet2Packet bridging SIP/proxy3-05ac9180 and SIP/proxy1-05acaae0 When someone dials in to 555828, I call two phone numbers, 1555111 and 1555222 simultaneously. The logging shows when one of those numbers is picked up, but I don't know which one. I'd like to be able to determine which phone number was picked up. How do I do that? Is there a variable somewhere I can tap in real time? The CDRs don't show which number was picked up either. Thanks! Enlai ___ -- 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] Determining picked up line from multiple line ring
I think I got it. ${DIALEDPEERNUMBER} contains the leg that connected
(just what I need).
FYI I used DumpChan() to get all the available variables and found it.
Thanks!
Enlai
On Wed, 24 Jun 2009 14:19:46 -0700, asterisk-users@lists.digium.com
said:
Thanks Danny.
I tried accessing ${CHANNEL} and ${DNID} or ${CALLERID{dnid)} in the h
(hangup) context, which is invoked after either party hangs up.
However, the ${CHANNEL} contains the original channel the call came in
on and not the outbound channel that connected. The ${CALLERID(dnid)}
contains the caller's phone number and not the one that connected on the
outbound leg.
Any other ideas? Should I put the ${CHANNEL} and ${CALLERID(dnid)}
somewhere else?
Thanks,
Enlai
On Wed, 24 Jun 2009 15:51:18 -0500, Danny Nicholas da...@debsinc.com
said:
${CHANNEL} or ${DNID} should do the trick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@enlai.net
Sent: Wednesday, June 24, 2009 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Determining picked up line from multiple line
ring
Hi all,
I've looked at the various variables but can't seem to find a way to
determine which line was picked up in a multi-line ring.
For example, in this excerpt from my asterisk logging:
-- Executing [5558280...@inbound:52] Dial(SIP/proxy3-05ac9180,
SIP/1555...@proxy1SIP/1555...@proxy1|18|r) in new stack
-- Called 1555...@proxy1
-- Called 1555...@proxy1
-- SIP/proxy1-05af5ca0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 is making progress passing it to
SIP/proxy3-05ac9180
-- SIP/proxy1-05acaae0 answered SIP/proxy3-05ac9180
-- Packet2Packet bridging SIP/proxy3-05ac9180 and
SIP/proxy1-05acaae0
When someone dials in to 555828, I call two phone numbers,
1555111 and 1555222 simultaneously.
The logging shows when one of those numbers is picked up, but I don't
know which one. I'd like to be able to determine which phone number was
picked up. How do I do that? Is there a variable somewhere I can tap in
real time? The CDRs don't show which number was picked up either.
Thanks!
Enlai
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[asterisk-users] Maximum cable length for analog phone from FXS port
Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- 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] Maximum cable length for analog phone from FXS port
Appreciate all your input folks. Much of it very helpful in the greater context of the initial question. Thank you for the suggestion of using various wireless devices, but I'm stuck with fixed wiring since this is a security/emergency phone(s) installation underground in large tunnels. Also, switching to VOIP is not really the answer here because then I'm forced to solve a lot of power, repeaters/switches problems that arise. So I'm actually worse of than using the analog connections I think. I do have some control over the wiring/cable chosen for this project but still forced to find a solution where I can feed the analog phone line the total 3km line distance. I would love to find a way to do this in the Asterisk context with some sort of FXS feed, either from Digium (or compatible) hardware or any of the available ATA boxes. The Sapura box suggestion may be something and I'll look closer into that as well as continuing to look for other ways to do this. tnx! Baldvin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 26. maí 2009 19:42 To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port I would suggest making a wifi connection with directional hi-gain antenna's. Ans a small box at the other end. Have a look at: http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit- pc.info/downloads/handleidingen/fit_pc_2_eng.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 ___ -- 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] br.Doctor Ester
About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the "Unsubscribe" link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. ©2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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 245058
About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the "Unsubscribe" link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. ©2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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] Best Sales 2008!
About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the "Unsubscribe" link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. ©2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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] Best Sales 2008!
About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the "Unsubscribe" link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. ©2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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] Sendmail for Voicemail
When I send email from my local asterisk machine, my IP address get's RBL'd. Asterisk is my only reason for running sendmail, so to keep it simple, I tried to make my ISP's mail server a 'smart host' (relaying to a trusted mail server) but my ISP doesn't allow ANY kind of relaying these days. I imagine there are many like me who are not sendmail experts who want to send Asterisk Voicemal. Can someone direct me to the quick, dirty and secure way to send mail from my asterisk box? The good news is that I'm on a Fixed IP on a registered network with working reverse in-addr.arpa lookups, and as you might have guessed, all mail would originate from the local host. Suggestions? Thanks! -Karl ___ -- 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] SALE 71% OFF on Pfizer
Dear asterisk-users@lists.digium.com, Best Price Only Today. http://byz.domemax.com?itg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best Practices: Empirical measure of call latency
I would like to hear your favored method to obtain an empirical measure of latency in the media path. I'm doing several things that bring the media path through asterisk, and this would allow me to make informed decisions about (a)PSTN termination providers (b)DIDs in local and remote locations (and variance between ITSP's) (c)time to/from various cellular networks (and variance between ITSP's) Thanks! Your opinion would be greatly appreciated -Karl Fife p.s. Speaking of latency, I've noticed that some sip endpoints (i.e. Aastra 57i Wireless) add significant latency. It would be interesting to do an apples-to-apples comparison between with various fxo/dect, sip/dect, wi/sip, fxo/Spread-spectrum digital , and fxo/analog 47/900/2400mhz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound video Calls
=no bindport=5060 bindaddr=0.0.0.0 videosupport=yes disable=all allow=ulaw allow=alaw allow=h263+ ;allow=h263 ;allow=h263p allow=speex allow=gsm #include /etc/pbx-tandil/sip.conf #include /etc/asterisk/sip_dps.conf [paul] type=friend username=paul secret=georgina nat=never host=dynamic canreinvite=no allow=h263p -- Paul Verity ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fw: Outbound video Calls
Hi, You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too. Maybe the switch wants to have it in Bearer Capability and LCC (I once had such a switch). Just applied the patch, failed again. can you tell me if theres anything more i need to add to the conf file to signal in LLC as well ? Another reason could be that the telco blocks video calls. They keep telling me that there shouldnt be a problem, however they are not the brightest bunch :-) regards klaus PS: use the asterisk-video mailing lists Just have :-) Asterisk Users schrieb: Hi all, I am trying to make an outbound video call to a mobile from asterisk. however it keeps failing. I can make inbound calls from a mobile and view video. I am using x-lite to initiate the outbound call, however I have tried using the management interface as well (action: etc...) and result is the same. normal voice outbound calls work fine. Circuit is a q931 30 channel from telewest (virgin media). Any pointers would be appreciated. below is pri debug output and relevant conf entries. // BEGIN // -- Executing [EMAIL PROTECTED]:1] Goto(SIP/paul-081ff260, video_test_out|666|1) in new stack -- Goto (video_test_out,666,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/paul-081ff260, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/paul-081ff260, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:3] h324m_gw(SIP/paul-081ff260, [EMAIL PROTECTED]) in new stack [Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't know any of 0x2000 formats -- Executing [EMAIL PROTECTED]:1] h324m_call(Local/[EMAIL PROTECTED],2, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CHANNEL(transfercapability)=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2, transfer=VIDEO) in new stack -- Executing [EMAIL PROTECTED]:3] Set(Local/[EMAIL PROTECTED],2, CHANNEL(userinformationlayer1)=38) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2, ul1=38) in new stack -- Executing [EMAIL PROTECTED]:5] Dial(Local/[EMAIL PROTECTED],2, Zap/g0/07525029025|40|tTkK) in new stack -- Making new call for cr 32771 -- digital call, setting user information layer 1 to 38 (0x26) -- Requested transfer capability: 0x18 - VIDEO Protocol Discriminator: Q.931 (8) len=38 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 88 90 a6] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: H.223 and H.245 (38) [18 03 a9 83 81] 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: 1 ] [6c 06 41 80 70 61 75 6c] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'paul' ] [70 0c c1 30 37 35 32 35 30 32 39 30 32 35] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ] [a1]CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call Initiated) -- Called g0/07525029025 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 80 e4 04] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 04 (4) -- Processing IE 8 (cs0, Cause) q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null) -- Channel 0/1, span 1 got hangup, cause 100 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:6] Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (video_test_out_context, dialcell, 6) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'UNKNOWN' == Spawn extension (video_test_out, 666, 3) exited
Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? ?In-Reply-To: [EMAIL PROTECTED] ?References: [EMAIL PROTECTED] [EMAIL PROTECTED] ? [EMAIL PROTECTED] ? [EMAIL PROTECTE
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Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
You are kidding, right ??? A small user that just buys one card won't get a good support from Digium. It'll be just a waste of time on the phone. Practically any manufacturer gives similar support including ssh'ing in the users box. Right now they push the user to buy a 4 channel echo canceller which you can get from Octasic for $40. The card with 4 ports is retail around $640. You can get OpenVox or another brand TDM400P compatible for 1/3 of that + $40 for echo canceller. Now that's a Digium high marigin right there .. someone has to pay the CEO salary and the mortgage for a new building :) cheers On 2/15/08, James Finstrom [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. ___ -- 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] Disable IAX2 call path optimization
I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller == Asterisk-A == Asterisk-B == Asterisk-A Now, what happens is that in my case both A and B are on the same network and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B out and the original caller talks directly to the extension hosted in Asterisk-A without the call path going the round-trip to Asterisk-B. Is it possible to prevent this optimization from happening? Any way to control if it happens at all, or can it be selected on per-call basis somehow? Can I find anywhere more details of call path optimization and it's configuration, use, functionality and behaviour? tnx, Baldvin ___ -- 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] Suppressing certain queue announcement voice prompts
[EMAIL PROTECTED] wrote: Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? Which announcements are you trying to not play? queue-thankyou for instance, to name one. Or any other of the queue-* files in general. From time to time it can be convenient to change the exact prompts played (order and contents) due to language differences and personal preference of the end-users. We're doing this now by replacing them with silence but I'm just thinking that it would be more elegant to have Asterisk not attempt to play them in the first place. We've also removed the files in some instances but that's even worse from my point of view because then we get file-not-present warnings. ___ --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] Suppressing certain queue announcement voice prompts
Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? Which announcements are you trying to not play? queue-thankyou for instance, to name one. Or any other of the queue-* files in general. From time to time it can be convenient to change the exact prompts played (order and contents) due to language differences and personal preference of the end-users. The question is more like what exactly do you mean with from time to time? Anyway, your best option is probably to create one or more prompt languages by copying the English prompts to a new directory like en2, en3 and then use Set(LANGUAGE=en3) in the dialplan when you think this is appropriate. For each of these artificial languages you can now decide how to modify the sound files. Cheers, Philipp Again, very good advice thank you Philipp. And probably a very reasonable way to do this if dynamic behaviour is needed. But in my case time-to-time was meant as every once in a while there is a particullar installation that requires this. So statically doing this is ok in my case. I'll continue with my replace-with-silence-file method for now. Thanks for the input. ___ --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] Suppressing certain queue announcement voice prompts
Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? ___ --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] Suppressing certain queue announcement voice prompts
[EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? Which announcements are you trying to not play? queue-thankyou for instance, to name one. Or any other of the queue-* files in general. From time to time it can be convenient to change the exact prompts played (order and contents) due to language differences and personal preference of the end-users. We're doing this now by replacing them with silence but I'm just thinking that it would be more elegant to have Asterisk not attempt to play them in the first place. We've also removed the files in some instances but that's even worse from my point of view because then we get file-not-present warnings. The sounds used are configurable in queues.conf. For instance, if you wanted to change queue-thankyou to play something else, you could add the line queue-thankyou = mythankyoufile inside a queue context. Unfortunately, the order the files are played in is not configurable. If you don't want sounds played at all, then there are certain options which you can simply not set inside a queue in order to not have the sounds play. If you don't set a periodic- announce-frequency, then periodic announcements will not play. Similarly, if you do not set an announce-frequency, then position/holdtime announcements will not be played. Well described and I understand that perfectly. The orignal point however was if it is possible to tell the queue application to not bother with certain announcements. I was hunting for some configuration options that are either not present in the queues.conf sample file or perhaps that I could find this in some totally different file that I may not have thought of already. Not because it's unclear how to replace them (as you described very well) with for instance a file containing very short silence or configure the queue so that they are not applicable (like the periodic announcement), but just to not spend time and resources on playing a file that we would rather not hear. Thank you for your clear reply though, you make an excellent point regarding the existing configuration options. ___ --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] Multiple B410P's in one machine
Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (although that was specifically geared towards their cards, I must say)? Thank you for your time and effort! Respectfully, Baldvin ___ --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] Building and running mISDN for B410P on Ubuntu 7.04
Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using make b410p I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using make b410P on Ubuntu? (make force does not help at all) 2) In some of our installations I'm getting stutter in the sound stream every two seconds or so (just under). I've tried to track this down to configuration but not been successful in spotting what the problem might be. Should I look for things like poll or dsp_poll values or does anyone have any suggestions that may help in pinning this? Btw, the setup is two B410P's in one machine, four ports in NT mode and four in TE mode. Feeding three ISDN BRI's into the system and three out again: PSTN - NT box - B410P port in TE mode - Asterisk - B410P port in NT mode - PBX Thank you for your time and effort! Respectfully, Baldvin ___ --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] Multiple B410P's in one machine
In an effort to better understand the interaction between multiple B410P's, mISDN, chan_misdn and Asterisk, I hope someone can add a bit more details to the clear and welcome answers presented so far. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: 17. nóvember 2007 23:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple B410P's in one machine [EMAIL PROTECTED] wrote: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? Yes, both possible and supported. Good to hear. Assuming I have the first card with four TE configured ports and four BRI's coming in and the second card with four NT configured ports and four BRI's going out to a PBX, am I right then in configuring this in /etc/misdn-init.conf like so (leaving out what I think is not really relevant to the discussion): ... card=1,0x4,rxclock card=2,0x4,pcm_slave ... option=1,master_clock ... poll=128 dsp_options=0 ... And in /etc/asteerisk/misdn.conf like so: [general] ... echocancel=yes echotraining=no echocancelwhenbridged=no bridging=no ... Incidently, if I set bridging=yes there is no sound heard between a call coming in on a TE configured port (no difference between PTP or PMP config) and going out again on an NT configured port. But that works (although with stutter in the sound every 1.5 seconds or so) if bridging=no. 1) Is this to be expected? Should Hardware bridging not be a better choice? 2) With hardware bridging, can MixMonitor still record the conversation? 3) Is there any reason one would not just use software bridging? tnx. ___ --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] Please explain the correct LED color for B410P
Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and then and then shining green when there is a call on one of the lines for that port. tnx, Baldvin ___ --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] Two B410P cards in one machine
Hi. I have two B410P ISDN BRI cards in one machine running Asterisk on Ubuntu 7.04. One card connects to the PSTN network and is therefore in TE mode on all four ports and the other card is in NT mode and connects to a PBX. The Asterisk is used to remap features, callerid's and more from the PSTN to the PBX. 1) Is there any special care I need to take regarding the configuration for these cards when they're put together like this? Especially concerning timing between calls bridged from one card to the other (PSTN call comes in, Asterisk answers it and connects to a new call going out on another port to the PBX)? 2) Is there a way to make sure that this is all run on the PSTN timing source through the asterisk box and over to the PBX? 3) Even though the call quality through the Asterisk box is ok as far as I can hear, I'm experiencing tiny drops in the audio stream at regular intervals (around every two seconds or so). My guess was timing slip of some sort between the cards or something like this, but perhaps I'm missing something that really needs to be taken care with when using two cards like this in one machine? Perhaps all the same question with a different twist, but I'm just trying to get the hang of this config and I can't find detailed enough documentation for this scenario via usual sources. All information relating to the correct or proper configuration of multiple B410P cards in one machine is very much appreciated. tnx, Baldvin ___ --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] How to get TCP access to CDR Master.csv
Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR's as they are coming in (being generated). I can't find this as a feature of the standard Asterisk... but maybe I'm missing something? The closest I could get is something around the manager api but it's not really what I'm after. I'd like to access the CDR's them selves. Being a (more or less) novice Linux user the only thing I can think of is trying to do this using Perl scripts where it would set up a listening socket and when connection is received it would do something like (in princip, not managed to do this properly yet): ... print $connection `tail -f /var/log/asterisk/cdr-custom/Master.csv` ... But even this is full of issues to solve. Things like only one connection at a time (which I can live with) from the remote computer. The fact that tail will not write to the socket (yeah, a major issue probably) which I'm thinking of trying to solve by reading line by line somehow and writing back to the socket... not even sure if this is possible. So basically I'm hoping someone has a nice solution for this. With or witout scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or whatever works. I'd really appreciate your input here. Sincerely, Baldvin ___ --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 TCP access to CDR Master.csv
#!/bin/bash while true; do tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l done Thank you John, this bash script is exactly what I was looking for. Very simple, yet works. As for doing this with insert into database and then polling for it... well I don‘t like polling. It‘s a good idea, but in the end, for this solution/in this case, the system reading the socket will in fact file the data (post processing) in a sql database for storing and querying. tnx, Baldvin From: John Hass [mailto:[EMAIL PROTECTED] Sent: 24. október 2007 22:39 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to get TCP access to CDR Master.csv ___ --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] H.323 trunk between MD110 and Asterisk
Hi. Anyhone have any experience with trunking between Ericsson MD110 and Asterisk using H.323? I've tried both ooh323 and the /channels/h323 one in version 1.4.4 and 1.4.0 of Asterisk. ooh323 does not manage to establish the call (starts to ring but then disconnection when answering the call on the Asterisk end) but using the channels/h323 driver I can get the call established from MD110 to Asterisk (still does not work in the other direction) but no sound is transferred between the two. Just dead silent on both ends. I have some logs and more details if needed and if anyone is ready to listen. Would really appreciate your input on this. tnx, Baldvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to tell what codec is used for each end of a call MD110-H323-SIP
Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you for your time and effort to respond. Baldvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Queue MOH
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TienSen Chong Sent: 17. maí 2007 10:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queue MOH Is there any way if i want the caller to hear dial tone rather than the MOH? Perhaps you could use something like Queue(yourqueuename|rt|||60); in extensions.conf or extension.ael? The r is defined as ring instead of playing MOH. Baldvin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correct setup for directing already ringing calls to newly available phones
[sorry if re-post - first mail to the list and did not get confirmation for
earlier mail]
Dear all.
Not sure what's the best way to describe the scenario and having searched
all over the place without luck, I hope some of you may have the correct
answer to this scenario:
I have an incoming PRI connection to Asterisk 1.4.2.
In the office we have two SIP phones and one Zap analog wireless phone.
Incoming calls are sent to these three phones in the dial plan
extensions.ael with:
Dial(Zap/67SIP/baldvinSIP/david/${EXTEN}, 50);
1) A new call comes in
2) All phones ring and the first one to pick up the handset gets the call.
3) A new call comes in.
4) The two phones NOT currently busy will ring.
5) The phone answering the first call hangs up.
6) I would not WANT the third phone to also start ringing... but this does
not work like that.
The problems I have are two:
A) The phone answering the first call does not start ringing when it becomes
available again. It only starts ringing if its free at the moment when the
call starts to ring.
B) Using the above way to route calls to the phones, the SIP phone NOT
answering a call registers a MISSED CALL. I would want the phone to know
that the call is not in fact MISSED but was handled by another phone!
Should any of you have any information to share regarding the best way to
configure Asterisk for this scenario, it is much appreciated. Thank you for
taking the time to read this and perhaps respond.
Sincerely,
Baldvin
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[asterisk-users] Choppy sound with chan_capi + Fritz Card USB
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[asterisk-users] custom sip header
hello, it is possible to include an particular sip header on outbound sip channels based on some particular conditions ? in particular I am interested to signal the context the call originated from to an on route sip proxy server. thanks, razvan radu ___ --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] GXP-2000 1.1.0.13 Issues
Grandstream have acknowledged that there is a problem with 1.1.0.13 on later phones (MAC's 00:0B:82:09:xx:xx I assume) and have advised me to wait for the next firmware release. So anyone with later phones (MAC's 00:0B:82:09:xx:xx), do not upgrade to 1.1.0.13. On Wed, 14 Jun 2006 [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 1.1.0.13 Issues
I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --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] GXP-2000 1.1.0.13 Issues
Thats what I thought the problem might be, so I have just now upgraded the other phone to 1.1.0.13 and its exactly the same, no speaker phone and hangs from a soft reboot. I also tried the audio loopback in the factory functions menu, this loopback's fine with the older 1.1.0.13 phones but does not with the newer ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx). -Drew- On Wed, 14 Jun 2006, Gareth Blades wrote: The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --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] GXP-2000 1.1.0.13 Issues
Thanks for the offer, but I have just tried 1.1.0.11, it is available publicly and it has the same problems on these 2 phones. On Wed, 14 Jun 2006, Mimmus wrote: If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works well with 1.1.0.11 firmware. I can send you this firmware, if you mail me off-list. Bye DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues Thats what I thought the problem might be, so I have just now upgraded the other phone to 1.1.0.13 and its exactly the same, no speaker phone and hangs from a soft reboot. I also tried the audio loopback in the factory functions menu, this loopback's fine with the older 1.1.0.13 phones but does not with the newer ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx). -Drew- On Wed, 14 Jun 2006, Gareth Blades wrote: The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Voice Prompts in Spanish
Anyone have or know where I can go to get the astcc voice prompts in spanish ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk over 3Com
I would if the tech that sets it up knows exactly what he or she is doing. Regards, Dovid : "Dakota" [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Asterisk vs 3COMTo: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; format=flowed; charset="iso-8859-1";reply-type=originalWould anyone recommend a medium size company choosing Asterisk over 3COM ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Decent sub-$100 SIP phone.
Ken, I would tell the client that you offerd phones for under $100.00 and he didnt like them so now for a diffrent phone he will have to pay more. Also I have an 841 and for it works great. I also installed one for a customer in a mechanic shop and no complaints. Regards, Dovid Message: 15Date: Mon, 09 Jan 2006 15:28:28 -0500From: Ken D'Ambrosio [EMAIL PROTECTED]Subject: [Asterisk-Users] "Decent" sub-$100 SIP phone.To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1Hey, all. I quoted a customer about $100 for some cheap SIP phones. Iwas planning on using the BT-102's, but he called said they look like"Princess phones," and I have to admit that he has a point. Some of theother inexpensive phones look decent, but (for example) the SPA-841'swiki entry says the remote end gets a lot of static. Since it'll bebeing used from a noisy environment (a cleanroom), the less overallstatic, the better. Someone suggested the Polycom 301's, but I'd losemoney on them. [I'll go with them if I have to, as I'm making moneyelswhere, but still...] So, does anyone have any suggestions for decentsub-$100, professional-looking SIP phones?Thanks!Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Mauricio, Yes it is. However I would not use analog phones. Your cheapest option would be to use softphones on a computer. If you wanted to use physical phones you have a few options. 1)Get two ATA's (device that you plug in to the LAN on your end and by your friend to the internet). This is probably the cheapest solution. You can plug in a "regular" analog phone in to the ATA device. 2)Use softphones that work on a computer 3)Get a TDM400P with one FXS port - this will cost a lot and your friend will need an ATA or VOIP phone on his end - This solution is howver worth it if you want to connect asterisk to your home line. 4)Get two VOIP phones. This sounds like the most sense. It will cost slightly more than ATA devices but they are much easier to use then POTS phones. Hope this helps and sorry if I am not to clear in the email A little tired. Regards, Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Jobs
Doug, I think that most companies dont know much about asterisk. Thier current PBX works for them and they believe in "if aint broken dont fix it". The only ones that seem to know about it are people that are currently working as IT or PBX people and they came across asterisk one way or another. (In other words it isnt really known out there yet.) Also a lot of people will be skeptical. I think the solutions is to brain storm and come up with ideas and sell it your self. I know for me and a friend of mine we are starting several small bussiness's based around asterisk. Eventually asterisk will get out there. When that happens companies will be scrambling for Asterisk tech's. The ones with the most knowledge will be us and it will serve as a benefit to us. Regards, Dovid PS Sorry about the spelling mistakes. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 18, Issue 46
Steve, I think that doug was new and didnt know how to act nicely. We taught him and he learned. We are all assests to this list. We are all human and we all make mistakes. What happend happend. We have to look forward from now on. David I am not sure why you are looking for jobs doing Asterisk work when less than two weeks ago you were publicly bashing on the list. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Client SIP fo Windows Mobile
Yes. I use X-Ten (now CounterPath) X-Pro for Windows Mobile devices. It costs about $30 and works relatively well on my Windows Mobile PDA. Note that you won't be able to readily use bluetooth headsets etc. but it works well enough using the internal speaker/mic on the device. That said, you may want to check with CounterPath's support to establish whether they support the Windows Mobile Smartphone Edition which has a different screen profile to the full PDA version. Regards Neil From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Monday, January 02, 2006 8:05 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Client SIP fo Windows Mobile Hi all, anyone known if is there any SIP client to install on an I-Mate SP5m with Windows Mobile ? Thanks Giordano ___ --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] Busy signal for incoming calls from broadvoice
Robert, The problem appears to be with your settings. I have an identical configuration with my * box running behind a NAT firewall with the same firewall ports open. I have experienced the same problem before. If port forwarding is switched on then do NOT use the nat=yes and externalip/localnet settings - this breaks it. If using regular Asterisk, suggest you copy the exact settings from here: http://www.broadvoice.com/support_install_asterisk.html If using [EMAIL PROTECTED], suggest you refer to these settings: http://voipspeak.net/index.php?option=com_contenttask=viewid=18 Merry Christmas everyone! Regards Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Saturday, December 24, 2005 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Busy signal for incoming calls from broadvoice trixter aka Bret McDanel wrote: On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote: When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? Did this start today or so? Rumor has it that BV is broken right now and others are having problems completing calls. I haven't been with BroadVoice long enough for my data to be relevent. i.e. less than 24 hrs. I tried calling their tech support and wasn't able to reach a live person and my call got dropped a few times. Haven't had problems otherwise. ___ --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] What hardware fits my needs?
Jami, Providing a specific response to your question is rather difficult without a more meaningful list of parameters. 1. You say you have 20,000 distinct DIDs already. Are these provisioned through an existing telephony switch using multiple PRI lines (E1/T1)? Ideally you would need 4 PRI lines to support the average load of 100 users (4xE1=120 channels or 4xT1=96 channels). However in reality there will be usage peaks - thus you may need to consider designing a system that can cope with double or treble that many simultaneous users in order to handle peak loads. 2. Providing that many mailboxes and offering the functionality you describe is feasible using Asterisk. However you will undoubtedly need multiple servers - though again the number of servers and their specification is dependent on many additional factors. 3. What is the nature of the service. i.e. Is it mission critical and do you need to ensure high-availability? This will impact the architecture/hardware configuration you choose. Also do you plan to locate all of the lines/servers at a single site or do you want to have redundancy spread across multiple sites in the event of an outage within your Central Office? 4. How many messages of what maximum length do you anticipate each user being allowed to store? Again this will impact storage requirements. 5. The www.digium.com site lists the cards they offer for interfacing to E1/T1 PRI lines. As for server hardware - you will ideally want to use fast multi-processor servers for your service. Again - the exacting specification is difficult to suggest without knowing more about what you are seeking to achieve. 6. Asterisk is robust and powerful. However there is a learning curve spanning anything from many weeks to a few months depending on your available skills/resources. Setting up a production grade service on this scale will certainly require a deep understanding of both Linux/UNIX and Asterisk. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of S.Ammad Jami Sent: Thursday, December 22, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What hardware fits my needs? Hello: I want setup an asterisk based VoiceMail Server(IVR). I have around 20K distinct users(DIDs) dialing to my system through telephones/mobiles. The users can dial to their mailboxes and listen/delete voicemails sent to them by others. The users can also recordsend voicemails to other users. I expect to have 100 simultaneous users to my system. Please suggest me the hardware configuration I need to have: the cards, peripherals, no. of extensions, hardware server etc. Thanks Jami __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] dtmf problem
Bart, We have has similar issues with BroadVoice in the past. From what I understand they had problems with DTMF depending on which proxy you register to. This is a bug that related to their session border controllers which should have been resolved. Looking at your config your first each account registers to a different BV POP (the IP address is of the first is different to the second account). Suggest the following course of action. 1. Try pinging each of the following BroadVoice POPs to find which is closest to you. proxy.nyc.broadvoice.com proxy.dca.broadvoice.com proxy.bos.broadvoice.com proxy.chi.broadvoice.com proxy.lax.broadvoice.com proxy.mia.broadvoice.com 2. Change your /etc/hosts file to so that the IP of the nearest POP from the list above is mapped to hostname sip.broadvoice.com NOTE: They did have problems with their LAX POP for DTMF. 3. Change your config file so all host entries point to sip.broadvoice.com instead of the 147.135.X.X IP address you're using at present! It make easier to make a single change to the /etc/hosts file for testing in future. 4. Change your DTMF mode from inband to rfc2833. BroadVoice does support out-of-band DTMF signalling, though their website is out of date. i.e. dtmfmode=rfc2833 5. You may want to check that your second register = statement is on a new line in your config file! ;-) We have out-of-band DTMF working properly across 6 separate BV accounts and register to their proxy.nyc.broadvoice.com POP without problems. As an FYI, BroadVoice also seem to unofficially support G729, iLBX and G726 codecs. If bandwidth is a problem I suggest trying out G726. We've been using G726 instead of G711u for the past week and have been impressed with the results Regards Neil/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Wegrzyn - asterisk Sent: Tuesday, December 20, 2005 8:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] dtmf problem hi everyone, I do have 2 lines with broadvoice. From 2 days on one line my dtmf tones are not passed to asterisk server. It siply goes through the extensions routine acting link it did not receive any tone. Could it be problem with my config??? It looks like this:(it worked for last 1.5 year) num2 is ok, but num1 is not working. Any ideas before I call support which is always a problem. [general] externip=lexon.ws bindaddr = 192.168.1.251 port=5060 localnet=192.168.1.0/255.255.255.0 disallow=all allow=ulaw register = number1:[EMAIL PROTECTED] register = number2:[EMAIL PROTECTED]/2000 tos=0x18 srvlookup=yes nat=never insecure=yes [sip.broadvoice.com] type=peer username=num1 fromuser=num1 authuser=num1 secret=pass1 host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [sip.broadvoice.com.home] type=peer username=num2 fromuser=num2 authuser=num3 secret=pass2 host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [broadvoice-incoming] type=peer host=147.135.8.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming2] type=peer host=147.135.0.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming3] type=peer host=147.135.4.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never thx Bart Wegrzyn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Teliax experiences
I have been using Teliax for several months now with no problems what so ever. However I did have problems with Broadvoice. The voice quality isnt allways that great. Sometimes the DTMF wouldt work. It was very frustrating when I dialed a company over my Broadvoice line and I tried to enter a number and nothing happend. Just my 2 cents. Regards,Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel
Dakota, Looking at it objectively, Asterisk has many benefits over traditional PBX systems, yet you should be aware of some of the limitations. Benefits: 1. Open source / low-cost of ownership / operates on cheap PC hardware. You get voicemail, IVR, hunt-groups etc. without additional fees. Last I checked those are all expensive add-ons in the Nortel world. There aren't expensive licenses per user/handset either. 2. Flexibility - you can configure Asterisk to handle calls to a microscopic degree of precision. This is just not possible with traditional PBX systems which are inherently proprietary. Asterisk also makes it easier to present data to callers from CRM, Billing, Order Tracking systems etc. using text-to-speech, automated-speech recognition and/or DTMF recognition. 3. Flexibility again - It really is much more flexible than anything else!! 4. Supports multiple VoIP protocols - SIP, IAX, H323, (and skinny to a degree) and supports connection of a broad spectrum of third party handsets - e.g. Cisco, Siemens, Sipura, etc. IAX is a proprietary protocol for Asterisk but it has some benefits over SIP (supposedly - my experience has been a little different) and perhaps more importantly is gaining popularity among VoIP service providers. Limitations: 1. Digium PSTN interface boards are not as cheap as they could be and haven't been around long enough for us to have meaningful data on how reliable they are. 2. Complexity. Asterisk is powerful but it is complicated - which is it You will need to spend a few weeks solidly learning about Asterisk and playing with it in a test environment before even thinking about trying to install it in a production environment. Clearly your time has a cost to your employer - thus this may be perceived as problem with Asterisk. You can of course buy in the services of an Asterisk consultant to help set things up - but ideally you want to have someone on site with some degree of knowledge about Asterisk's capabilities. If your business has basic telephony requirements, doesn't need fancy features and wants to minimize the need for on-site technical expertise to support Asterisk, then a Mitel/Nortel solution MIGHT make sense. IMHO - the present level of complexity/flexibility is the biggest strength and weakness to Asterisk. 3. Asterisk is a work in progress. Yes it's pretty stables and yes it's being used in very large production systems from what one hears on this list. However it's a moving target with new releases appearing frequently. On a positive note that's great if you want new features and bug fixes - but it can also be a pain if you want a nice stable, low-maintenance system. 4. Cost savings aren't necessarily as great at they first seem. You ideally want to have redundancy on your Asterisk set up. To support 75 users you probably want to have a couple of decent Dual-proc Pentium Xeon servers. Sure you can build these cheap - but if your company is like mine you'll probably buy from Dell/HP etc. which can make that a not-insignificant investment. Then you'll need 2x PSTN interface cards for each machine. Depending on your PSTN lines there this can cost anywhere from $800 - $3000 per card. So overall you can be talking perhaps upwards of $10,000 for the hardware to support your asterisk installation. Handsets would obviously cost more though you have the flexibility to choose any pretty SIP/IAX handsets you like. Hope these observations help. N -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Tuesday, December 06, 2005 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel How does Asterisk compare to Nortel, NorthStar and Mitel PBX systems? For a medium size company not growing past 75 extensions, would you recommend Asterisk? ___ --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] Recording voice messages in mp3 format
You'll find the GSM codec renders smaller filesizes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil Sent: Wednesday, November 16, 2005 12:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recording voice messages in mp3 format Hi, Is there a way so that I can record the voice messages in mp3 format instead of wav? I think it is much smaller in size compare to wav. It is also easier to send small sized file as an attachment. Currently when my users record voice messages the format is wav. Where can I configure it so that it will become mp3? Thanks, Ryan ___ --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
[Asterisk-Users] Echo
I've got a customer on an IAXy and another with their own Asterisk box as a PBX with an array of Cisco, GrandStream, ATCOM, and xten hard\soft phones. Same LEC, same Asterisk box on our end, same broadband provider on the client ends With no packet loss, 15 ms pings, 13 hops, the IAXy sometimes has an echo, some times not. The client with the Asterisk box... no problems at all. What could I do to figure out what's going on here? --Mike This message was sent using IMP, the Internet Messaging Program. ___ --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
[Asterisk-Users] Privacy Manager Application
Hello, I am trying to utilize the PrivacyManager application to request entry of a CallerID value before allowing a caller to enter an IVR menu. The documentation at : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManagerwas very limited. Also from what I can see, PrivacyManager will not stop calls that are labeled "anonymous" by my service provider. Does anyone have any sample code and perhaps a solution to the second part of my question? Many thanks in advance, Neil ___ --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
[Asterisk-Users] SNOM 360 Unknown SIP command 'PUBLISH'
Hi List Im getting this notification from my one and only SNOM 360 every time a number button is pushed. I know that its only a notification, but it really irritates me. Is it anything I can/should do anything about ?? Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530 handle_request: Unknown SIP command 'PUBLISH' from '192.168.100.100' By the way Im using * 1.0.9 CVS-HEAD September 15. 2005 Best regards BennyBad ___ --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] hints and the sNOM 360
Hi Paul It's working for me ! (CVS-HEAD 1.0.9 FC3) I'm using the snom 360IP with firmware 4.2 http://www.snom.com/download/snom360-4.2-SIP-j.bin In my extensions.conf I have: exten = 100,hint,SIP/100 ; SIP Phone 100 exten = 101,hint,SIP/101 ; SIP Phone 101 exten = 102,hint,SIP/102 ; SIP Phone 102 On my phone I used the same setup as You. A good hint is: Be patient. It often takes up to 5 min. before it starts working for me. Normally I start *, start snom, start other phones. Hope this is of any use ! Reg. BennyB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hewlett Sent: 19. september 2005 18:49 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] hints and the sNOM 360 Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on 'hints' and also on the 'devstate' app. I set the first function key on the 360 to extension 2001 - this transforms itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key type is set to 'Destination' as recommended by a number of articles on the Wiki. aside This seems to contradict the 360 manual which states that the function key type should be set to 'Line'. /aside In the dialplan I put [myhints] exten = 2001,hint,sip/2001 exten = 2001,1,macro(stdexten,sip/2001) exten = 2001,2,hangup In sip.conf I have [2001] type=friend username=2001 subscribecontext=myhints host=dynamic mailbox=2001 callerid=ext 2001 incominglimit=1 [2002] type=friend username=2002 subscribecontext=myhints host=dynamic mailbox=2001 callerid=SNOM360 2002 I restart asterisk from scratch and then reboot the 360. The * console shows one entry when typing the command 'sip show subscriptions' which looks correct. Inspection of the sip trace log on the 360's web page reveals that the registration succeeds and that the subscription of the 2001 from the 360 also gets a 200 OK reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --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
[Asterisk-Users] Submitting ISDN-MSN from a SIP-Phone
Hello,
i wonder why i didn't find a solution for this problem yet, because it
seems very common:
I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some
SIP-Softphones which i can call from outside by calling the phonenumber of the
Asterisk-Server and then dialing the number of the SIP-Phone.
If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server
is submitted, without the extension of the SIP-Phone.
I tried to give Asterisk several MSNs in capi.conf and to dial in
extension.conf like the following:
exten = _0.,1,Dial(CAPI/@ASTERISK-MSNSIP-EXTENSION:${EXTEN})
This was just for testing, so i used a fix SIP-EXTENSION, but just
the ASTERISK-MSN was submitted.
Is there a way to submit the whole number ? Is it generally possible to do
this with a BRI-Card ?
Kind regards,
Holger
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[Asterisk-Users] ZyXEL P662HW / SIP / Crashing
Has anyone experiences this please: - We were running a number of ZyXEL P662HW-61 routers at our sites and all traffic was being sent over IP-SEC VPN's between devices. When we moved to a new architecture, we got rid of the VPN links so that the SIP traffic was running directly through the routers. Each site uses Snom 360 devices with the latest firmware (v4). Ever since we did this, the routers have been crashing at least 5 times per day. They appear to carry out a full cold start each time (as though they are having a kernel panic). The ISP is Nildram in the UK, but we have also experienced this a few times with another router in France on a France Telecom system. As soon as we route the SIP traffic via another router, stability returns to the network. Our supplier has been very helpful and we have tested every release of the firmware from the last 8 months, but they all behave the same once SIP is being transmitted. The routers are running with their most basic configuration now, but this doesn't appear to make any difference. Does this sound familiar to anyone please? We are out our wits end and our supplier has no ideas (and neither do ZyXEL it would appear). For reference, all traffic is being sent through using G.729, but I don't think that this makes any difference. Many thanks John ___ --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
[Asterisk-Users] SIP Connection Problems
Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 1-2). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David ___ --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
[Asterisk-Users] Computer to use
Hi, Already posted once but I need more feedback. What kind of servers is everyone using for asterisk and what problems have you ran in to ? Thanks.Dovid ___ 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
[Asterisk-Users] Kind of Computer to use
Hi,I am building PBX's for clients. I was thinking of using Dell computers. I was told that they do not work well with asterisk. Any one have any suggestions ? Any other brands that work well with asterisk ? Also any specific hardware to or not to use ? Finally does that Mac Mini work well with asterisk ? Thanks a lot. Dovid ___ 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] Programs to parse queue_log
Hey Johann! Just thought I would mention our upcoming 2.6 release of PhoneCALL. We already have routines in there that check system logs, and Asterisk logs. You can check it out at: http://www.vecsector.com/phonecall Click on the DEMO on the right-hand side. User/Pass: demo/demo Look on the bottom left-side under 'Logging', then 'Phone System'. I just added the 'Queue' section for you to see. :-) Enjoy! --Dustin Wildes Johann wrote: What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann ___ 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 ___ 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
[Asterisk-Users] Router with QoS recommendations
Hi List As I have a Cisco PIX 515, with NO QoS functionality, and Im looking for a router that does outgoing QoS to put in front of my PIX. Problem is that Im using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data is being sent to the internet the sound quality drops to something that is of NO use. Any suggestions or recommendations is appreciated. Best reg. BennyB ___ 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
[Asterisk-Users] InterVivo and MusicOnHold()
Hi All, I've been trying for a while to get * to play MusicOnHold with my SIP connection. I can hear it when I call a test extension from my local X-Lite phone, but when I dial in via InterVivo, I just hear silence. I have a Gentoo box with kernel 2.4.28-gentoo. I have no sound card or speakers on the box, it's in a cupboard. I have uncommented the lines in musiconhold.conf. I am trying to use the following extensions [inbound-calls] exten = s,1,Dial(SIP/07X,20,m) [voip] exten = test,1,Goto(inbound-calls,s,1) Dialing test from X-Lite works correctly, and dialling in diverts to the mobile, but with silence. PlayBack() with GSM files works okay. Is there something special I need to do with InterVivo to get it to work? Thanks Jamie SIP.CONF attached: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = inbound-calls ; Default context for incoming calls srvlookup = yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc externip = register = 0207043:[EMAIL PROTECTED] realm = voip.project76.net localnet = 41.0.0.0 localmask = 41.240.0.0 nat = yes ;outbound calls go here [sip-with-london-number] type=friend secret= username=0207043 host=sip.intervivo.net insecure=very fromuser=0207043 fromdomain=sip.intervivo.net ;soft phone client [jamie] type=friend secret= host=dynamic nat=yes username=jamie disallow=all allow=gsm allow=ulaw allow=alaw context=voip -- Visit our Little Britain microsite: http://little.britain.project76.tv/welcome.php You can now contact us at local call rates(*) via our NEW number: 0845 226 9157. (*) May not be included in your provider's call allowance. Check with provider for call costs. ___ 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
[Asterisk-Users] Soft keys and transfer problem on Sayson 480i
I have a strange problem with my Sayson 480i IP phone. If I press the Transfer button and then dial extension 200 to try to transfer the call, the Sayson apparently is treating the 200 as the last part of an IP address, and the call fails. As soon as I enter 200 and press Dial, I see an IP address of the form x.y.z.200 show up on the display. How do I get the phone to treat the dial string as an extension to be processed by the PBX rather than an IP address? If I simply pick up the phone and dial a 3-digit extension it works, but not when I try to Transfer to a 3-digit extension. Secondly, I have no documentation at all on how to program the 6 soft keys. It's not explained in the User Manual. Anyone know how? Thanks. -Ron ___ 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
[Asterisk-Users] VoiceConduits - Notice
Hello, This is David Deutsch, and Im the owner of VoiceConduits. There seems to be some confusion related to our company, regarding the past few posts. VoiceConduits is currently NOT open for public business, we have never to date advertised or attempted to attract business. It appears that a few people heard about our company via a mention in a SineApps article and found our beta system that is under development. We apologize that a few people managed to sign up via this interface, and we will happily refund anyone who did so immediately, additionally we will supply them with free credit to be used once we are in fact live. It was certainly never our intention to defraud individuals of the asterisk or voip community, our understanding is that only 5 people have managed to signup thru this automated system, and we will be contacting each of them individually to insure they are refunded and happy with the resolution. Thank you, David Deutsch, President Tris Telecommunications, LLC (800) 547-4057 x1001 ___ 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
[Asterisk-Users] Cisco 7960 Support Products
I just tried to order the CON-SNT-CP7960 part from CDW This is the ~$8 1yr support contract thats supposed to give access to the Cisco download site for firmware for the 7960 we, I got a call from CDW saying that Cisco wouldnt authorize them to sell that product to me. The sales rep conferenced me in with the CDW Cisco person and he explained that CON-SNT-CP7960 is an international product and that the equivalent U.S. product was CON-SNT-PKG1 which is a catchall support contract for all Category 1 Cisco products (which the 7960 apparently is) and costs $86.97. I asked him why the domestic (U.S.) version of the same thing was 10 times more expensive and he replied that there usually isnt much of a price gap and he had no idea, he also mentioned that hes been trying to figure out what the hell Cisco is up to for the past three months. He also mentioned that other resellers might be able to sell CON-SNT-CP7960 in the US, but that CDW was not authorized to do this. Now, paying $86.97 for a year of phone support isnt really that big a deal for me, but since all I (we) want is firmware access on the Cisco website, it seems kind of ridiculous. I was wondering if anyone else has recent experience getting CON-SNT-CP7960 in the U.S.? What retailer did you use? Weird. ~Adam ___ 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
[Asterisk-Users] Small PBX to VoIP transition questions
This is for a small business (restaurant and catering). We want to move from POTS to VoIP to save on the phone bill. Currently we use four lines for voice + one fax going into a Lucent Partner system PBX. Right now I'm considering two alternatives, both powered by * and a SIP/IAX wholesaler (any recommendations on that for someone who can do LNP on DIDs in Boston, MA, USA area are gladly accepted). Alternative One) Use a Digium card in the * box to drive the PSTN lines going into the Lucent system. This is the simplest and cheapest alternative, as it leaves all the current phone equipment in place. From what I've read I think it shouldn't be a problem to get incoming calls on the VoIP to hunt through the FXS interfaces. Alternative Two) All VoIP: buy new phones (probably Cisco or Polycom IP phones), a PoE switch and some ATAs. My hesitations in this area (aside from the cost) are mimicking the functionality of the partner system. Because this is a restaurant environment, there are only three phones that will be used as Office phones -- the rest are floor phones. In the partner system these are the cheapest phones offered: 4 button, no display. On these floor phones, the four buttons are just used as line buttons. Incoming calls always ring all phones. A manager can answer a call on the floor, put it on hold and return to the office. Incoming calls for employees are put on hold and the page feature is used to let them know (i.e. Bob, call for you on line three). It wouldn't really work to transfer the call to a specific extension, since the workers move around and need to be able to pickup an arbitrary line from anywhere there is a phone. All the lower end IP phones I look at say they have two lines ... I'd really like to use cheap(er) phones on the floor: they get abused and gross and don't need to do much (except be four line phones) -- and need to be replaced more frequently than normal. I'd like to hear other's ideas on how to implement this system or if others have implemented VoIP with floor phones in a restaurant, warehouse, etc... Finally) Would I be foolish to try to send / receive fax over VoIP by plugging the fax server into ATAs and using a zero compression codec? We send about 150 faxes daily and receive a couple as well...having to keep more than one analog line around for FAX would defeat the cost saving motivation behind all this anyway (we can theoretically use up to three of our five lines for FAX at a time). Internet Fax is so overpriced its absurd (unless someone knows of a company doing it for around $0.02 a page, which is how much the phone call costs to send a one page fax). Thanks for making it through my long post...I'd by happy to get any answers about any part of what I mentioned above! ~Adam ___ 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] four wildcards in a single pc
On Mon, Dec 13, 2004 at 03:15:18PM -0600, Grady Trew, Jr. wrote: Getting dedicated IRQs for the cards is a minor problem compared to what happens when you have four cards hammering away mercilessly at the chipset and CPU of your motherboard; 1000 IRQs per second, per card. Nobody's really sure what's wrong, but it causes problems for pretty nearly everyone. Would the same issues arise with the use of a single Voicetronix 12 port card? What about using 2 of them in the same machine? This would be rather silly as you'd be better off price wise getting a single T1 card and a channel bank. Otherwise, the Voicetronix board may or may not even do the 1k/s. -- Mike Mattice - Systems Programmer and Administrator ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] feature suggest.: alt. include criteria
Hi guys, I've got a quick feature suggestion to solve a problem that I don't think is readily solvable having to do with an after hours message, playing only during (or rather outside) specified times. I know all about the helpful feature that already exists which allows you to tack on a cron-ish specification of time, date, month, etc. to the end of an include = statement to conditionally include it. However, I've been thinking about a feature that'd be great (and possibly not difficult to code, though probably beyond my immediate Asterisk coding skills) for helping to implement an after hours w/ optional override feature, similar to what this poster mentioned: http://lists.digium.com/pipermail/asterisk-users/2004-November/071460.html I think it'd be great if we had the option to conditionally include contexts based *either* on time [the existing capability] *or* based on a general Boolean expression [just like the capability in GotoIf's, e.g.] or at least something similar to that power, where you could examine the contents of a GlobalVar and decide to include a context based on the variable. Am I correct in my understanding that this currently does not exist? Does anyone else think this is a good idea? Has it already been suggested (I didn't exactly find a reference anywhere yet)? For the record, I should mentioned I appreciate there are many ways to solve this with existing capabilities, and I have managed to hack together a solution that mostly works. I was just a bit wistful while mucking up my previously neat looking extensions.conf with some not-so-clear code to handle this. (I ended up using a combination of Macros GotoIf/Times). It seems to me that what ended up taking me a 5-10 extra lines of code, and still doesn't give me perfectly what I want would be trivially solved (with like one line of extra code) if we had the feature I'm suggesting. Thanks to all the developers for a wonderful, flexible, robust Open Source PBX solution. John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cisco 7940 help
Hi, I spent 3 days trying to upgrade my 7960's with similar probs to what you say. My problem was that my TFTP server didnt follow the RFC's properly and didnt respond with 'file not found' when a file didnt exist. the phone kept on looking for the same file over and over and never moved on to the next step in the upgrade process. the trick for me was to use a different TFTP server. Klever Pumpkin, Windows 2000, and the newer 3Cd all worked. my original broken TFTP server was a 3Com one. so perhps you should try pumpkin as its prolly the easiest for a temp solution. cheers, Mick Andrei (MPI) [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Rich Adamson wrote: I've successfuly converted 7940 from call manager firmware version 3 to SIP 7.3. just last week. You need to upgrade to firmware version 6 first, then upgrade to 7. Also once you've upgraded the phone, you should remove firmware config file from tftp server, otherwise the phone would be in constant upgrade loop. There is couple of tricks in between. The phone does a version check before attempting an upgrade. If the same, it doesn't bother upgrading again. No need to remove anything. In fact, my server has v2.3, 3.3, 5.1, 6.1, etc on it at all times. Some of those are required (in steps) to upgrade the older 7960's. Rich, I am telling what I saw on my 7940: it was in a constant loop trying to upgrade again and again. You may say that Cisco phones are easy upgradable and all one need is just follow instructions, but that won't help the man who was pulling his hair off. Andrei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Japanese FXO card
Hi folks, Im totally new to * but I went ahead and told my boss that it was the way to go for our new telephone system :) now I have a test box and two cisco phones and a brand new modem card. Im having plenty of trouble with learning all the config stuff but ill leave that for another day. ie: a few days after I rtfm. My modem card, once installed in the box (FC2 by the way) was detected by linux and installed perfectly no probs. but now I dont know how to make * recognise it? How can I tell if it is even compatible with *? I dont think all the usual options of buying the compatible cards are open to me because im in Japan. We have a bunch of ISDN lines and TAs to use, so if im out of luck with the modem I bought perhaps Id have a better chance with an ISDN card? your thoughts/comments/suggestions are appreciated. cheers, Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk
After a recent upgrade to asterisk HEAD, my asterisk startup scripts don't properly start asterisk. They have since May, which is the last time I upgraded. I am on Slackware 9.1, running kernel 2.4.26. After reboot, lsmod shows wct1xxp, then zaptel, which would indicate it now loads out of order? Shouldn't zaptel be loaded first? Maybe my original install is a little hacked. Where do you load all your modules and asterisk from on startup of your server? I have a T100P and a TDM400P installed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Asterisk Consultant visiting San Diego
Dear All My business provides Asterisk consultancy in the UK. I am traveling to San Diego / Tijuana from the 4th to the 13th and wondered if there were any fellow Asterisk users who would like to meet for a coffee / drink? Please email me direct ([EMAIL PROTECTED]). Regards John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceXML / Asterisk
Dear All Is there anyone out there who is using a VoiceXML system with Asterisk? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Connection to Splicecom Maximiser
Hi Everyone We would like to connect our Splicecom Maximiser PBX to our Asterisk box via H323 so that we can send our US calls via a low cost carrier (e.g. Broadvoice). Has anyone managed to do this in the past (I remember seeing some companies also worked with this system in the UK). The Maximiser only speaks H323 (not SIP) and can act as an H323 Gatekeeper, so in theory we should just be able to log on to the system (he said, wishing it was that simple!). Many thanks John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling on Mac OS X (10.3.5)
Dear All I have successfully used the packaged version of * on the Mac for some time, but decided that I would recompile one of the more recent builds so that my PC and Mac were in sync. As suggested, I installed the XCode tools, updated bison and downloaded the latest version of *. Unfortunately, when compiling there are lots of errors, many of them relating to the non-existent /usr/src/linux directory. Is there something special which I should be doing? Thanks JB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kernel 2.6 and zaptel data
I saw somewhere that the last kernel to work properly with the zaptel drivers when using data over it was 2.4.20. Has this been since fixed to work with newer kernels? -Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming cid translation tables
How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as 429102 which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix prepended or 1-393-942-9102. Is there some simple way to route all the incoming FWD calls to a context that prepends 393-9 to their 6-digit prefixes? (And for extra credit 393-99 to their 5-digit prefixes?) Unless I can fix up their CID, the dialback buttons on the phone (and in voicemail) are useless. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Send personal replies to this address. Mailman won't let me post unless I forge the From-line to be the same as my incoming alias for this list. [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Discriminate on IAXTEL dial-in
Hi, I have two IAXTEL accounts, which I activate with: register = alphanet:[EMAIL PROTECTED] ; 1-700-895-5211 register = cril:[EMAIL PROTECTED] ; 1-700-669-1152 when someone dial this number, it goes through the iaxtel-user context. In extensions.conf, I tried: exten = 17008955211,s,Goto(iaxtel-guest,s,1) exten = 17006691152,s,Goto(isdn-free-dial-out,,s,1) unfortunately it doesn't seem seem to work easily, maybe because IAXTEL doesn't send me the called ID ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
