Re: [asterisk-users] Callwithus.com is discontinuing IAX service
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote: Callwithus.com is discontinuing iax service. Can anybody recommend IAX provider - I need somebody with good rates to Philippines. -- Joseph ___ -- 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 Voipstreet.com seems to have very good service. I have been testing for several weeks now, without issue. -- A.G. (Tony) Nichols I.S. Manager ___ -- 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] * + Legacy PBX works but strange problem
On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Sun, Nov 16, 2008 at 4:28 AM, Sriram [EMAIL PROTECTED] wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)-
legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which is in
sync with asterisk on span2)This works perfectly fine until about 200
calls or so...After that time when asterisk tries to dial to the legacy pbx
- the call drops with error All are busy congested at this time .the same
is indicated on asterisk -rvv , but the spans are up and active at
that time... can anyone throw some light on this ?
ZAPTEL.CONF
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
bchan=32-46
dchan=47
bchan=48-62
ZAPATA.CONF
context=pri-pstn
switchtype=euroisdn
pridialplan=local
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=1
pickupgroup=1
immediate=yes
musiconhold=default
signalling = pri_cpe
channel = 1-15
channel = 17-31
context=pri-legacy
immediate=yes
group=2
overlapdial=yes
signalling = pri_net
channel = 32-46
channel = 48-62
EXTENSIONS.CONF
;
; Context PRI-Public
;
[pri-pstn]
;
include = default
;
exten = s,1,Answer
exten = s,2,Dial(Zap/g2/1888); Dial to legacy pbx and sends the 4 DID
digits needed for the legacy pbx
exten = s,3,Hangup
;
; Context PRI-legacy
;
[pri-legacy]
;
include = default
;
exten = s,1,Answer
exten = s,2,DigitTimeout,2
exten = s,3,ResponseTimeout,2
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
This is just a suggestion that has worked very well for me in the past when
dealing with Legacy systems that have only Analog phones connected.
Ditch the Legacy system and get some form of channel bank. If you want to
go SIP to Analog, I have had great luck with Quintum Tenor AX. Since, you
have a spare E1 port, you could simply terminate the analog lines to a tried
and true channel bank. I have never looked for an E1 channel bank (30 port
density) but I would assume they exist.
If the Legacy system has proprietary, digital extensions, that complicates
things a bit.
Special apps running or connected on your Legacy system can usually be
migrated and after that bit of growing pain, you have all the flexibility
you want to customize.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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I have noticed when connecting our legacy system to asterisk, the option
overlapdial=yes
caused issues with only certain exchanges... and would appear randomly. It
seems to add a pause of some 4 sec. when dialing.
This would give you the busy error.
--
A.G. (Tony) Nichols
I.S. Manager
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Re: [asterisk-users] 64 bit HPEC modules available?
On 2/16/07, Greg Siemon [EMAIL PROTECTED] wrote: I am running 64 bit linux on my Asterisk box and would like to get the new HPEC software running on it. However, while there are 32 bit modules available, there are no 64 bit modules on the ftp site: http://ftp.digium.com/pub/telephony/hpec/64-bit/ In some places on the digium website it states 32 bit only and other places including the documentation it states 32 64 bit are available. Is there a 64 bit version of the HPEC module or are the 32 bit modules suitable (can't imagine that they would be). Thanks in advance Greg ___ --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 I talked to tech support today... no 64bit yet. -- A.G. (Tony) Nichols I.S. Manager ___ --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] FYI new aricle on asteisk
Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network. I've also been in contact with Novel/SUSE about their asterisk pakages. Reinhard Max the maintainer. He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is unsure about the new zaptel intergration but I'm keeping my fingers crossed! -- A.G. (Tony) NicholsI.S. Manager ___ --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] SUSE 10.1
I have been told the next version of SUSE will contain the 1.2.1 build. I am unsure if the zaptel module will be ready -- but I have hight hopes! Per my last post... 10.0 is working very well in production -- including the auto updates.-- A.G. (Tony) NicholsI.S. Manager ___ --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] Tiny Echo on PSTN via Zaptel
I have had problems between the sip/FXO lies and was able to kill the echo by trying different combinations of the echocancel line to 64 (I think it has settings in 32 bit increments) Just kept trying different ones till it went away. Here is my config: group=1 context=line1 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 1 context=line2 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 2 musiconhold=default context=line3 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=64 echocancelwhenbridged=yes callgroup=1 rxgain=1.2 channel = 3 group=2 context=line4 signalling=fxs_ks usecallerid=yes callerid=asreceived echocancel=96 echocancelwhenbridged=yes callgroup=2 channel = 4 Hope this helps! t o n yOn 9/28/05, Shaw Terwilliger [EMAIL PROTECTED] wrote: I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhinochannel banks (one 12FXO/12FXS, the other 24 FXS).So it's an analogphone on the inside connected to one of the FXS ports, and PSTN lineconnected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial outthrough the PSTN line, I hear a tiny echo on the phone (I estimatebetween 20ms and 40ms), which never goes away for this call.It's just loud enough to bug the heck out of me when I'm talking (I could estimatethe gain relationship with ztmonitor if it would help).The soundon the recipient end of the connection is perfect.If I make a call from the phone to the another internal extension (another phone on an FXS port), there is no echo.If I call into Comedian mail,there is no echo.I've checked all my gains.The internal gains were a bit loud to startwith because of the powered phones, but now they all fall comfortably within ztmonitor's dynamic range display.The PSTN line is pretty good at tx 0 andrx 0, so I left it.I've tried turning them down, but that didn't killthe echo.My zapata.conf includes these lines at the bottom: echocancel=yesechocancelwhenbridged=noechotraining=yescontext=companyA-pstntxgain=0.0rxgain=0.0signalling=fxs_ksgroup=1channel=1-7context=companyB-pstntxgain=0.0 rxgain=0.0signalling=fxs_ksgroup=2channel=11-12context=internaltxgain=-12.0rxgain=-8.0signalling=fxo_lscallerid=asreceivedgroup=3channel=13-48When the calls are connected, I can use zap show channel 11 and verify that the echo cancellation is ON.But I can still hear one.I've also tried echocancelwhenbridged=yes, but it didn't make any difference.--Shaw Terwilliger [EMAIL PROTECTED]SourceGear LLC-BEGIN PGP SIGNATURE-Version: GnuPG v1.2.5 (GNU/Linux)iD8DBQFDOuxyPEbgvbl6u4ERAh/gAKCAu+gAr9TYsMG5TYqozV3ebIvezwCdGIHkkftrFLFK4purux/sVIPRhKk= =9WG3-END PGP SIGNATURE-___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A.G. (Tony) NicholsI.S. Manager ___ --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] Zaptel Problems with 1.0.9
On 7/27/05, Paul Traue, Jr. [EMAIL PROTECTED] wrote: I'm experiencing rather severe problems with 1.0.9 (we've had to backrev to our last version we know works (1.0.5). We are running a single PRI line with a T100P card. After about 10 hours of asterisk running and the modules loaded we start hearing noise and stuttering on any call that passes over the PRI line. I've tried this with echo cancellation on and off with no difference. This is a new problem for us as 1.0.5 behaves perfectly in this regard (it has it own issues, but that's another story). We would like to move back to 1.0.9 however restarting out phone system (which is in production) every 10 hours isn't really an option. Is anyone experiencing any similar symptoms, and if not what information would the developers need to work on this. Please note that running unstable isn't an option as the only PRI line I have to play with at the moment is our main line. Paul I have it in production on 8 servers (2 have pri's) no problems. One remote site keeps crashing when I do a compile so I left it at 1.0.7 The other 6 are hp d220's with fxo's installed. -- A.G. (Tony) Nichols I.S. Manager ___ 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] Trying to get *8 call pickup to work
|alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:7329 handle_request: Check for res for 1310 Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:1620 update_user_counter: Call from user '1310' is 1 out of 0 Looking for *8 in default Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:4650 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED] list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08 From: Test SIP sip:[EMAIL PROTECTED];tag=5eba6d75ff7e1e47 To: sip:[EMAIL PROTECTED];tag=as23dd6dfb Call-ID: [EMAIL PROTECTED] CSeq: 48201 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to pickup.phone.ip.addr:5060 Jun 28 10:43:23 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 10:43:23 NOTICE[16774]: chan_sip.c:7402 handle_request: Nothing to pick up Reliably Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08 From: Test SIP sip:[EMAIL PROTECTED];tag=5eba6d75ff7e1e47 To: sip:[EMAIL PROTECTED];tag=as23dd6dfb Call-ID: [EMAIL PROTECTED] CSeq: 48201 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to pickup.phone.ip.addr:5060 Please also let me know if any other information would help to troubleshoot this. Robert Woodcock Sr. Network Engineer Print, Inc. (425) 629-2424 http://www.printinc.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 -- A.G. (Tony) Nichols I.S. Manager ___ 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] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
On 6/19/05, Bob Goddard [EMAIL PROTECTED] wrote: On Friday 17 Jun 2005 18:05, Manuel Casal wrote: Marco Parmeggiani escribió: Manuel Casal ha scritto: I made the make menuconfig and make dep in the kernel sources. i do not remember well how i solved that problem but i'm sure that make dep will issue you a warning and stop. run make to start the kernel build process and then stop it after few seconds. it will create the necessary symlinks in the kernel tree. maybe there's a more elegant solution but this should work. [... Yet another f*ck*ng signature not deleted ...] [...] SUBDIRS=/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make: *** [linux26] Error 2 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Now what?:( Have a read at /usr/src/linux*/README.SUSE I never could get 9.2 to work right... even with THEIR asterisk rpms (from the install cd) However they seem to have fixed it in 9.3 (I have 3 installations working well). Digium tech support won't touch either of them. They told me SUSE changes a bit somewhere that prohibits loading modules that don't have it. If you don't want to use the 9.3 rpms you may have a problem on your hands. sadly I realy like thair distro. wish they would let us know how to make non suse modules work. With 9.1 I followed the admin guide; and it always worked: cd /usr/src/linux zcat /proc/config.gz .config make oldconfig Then make the symlink for zaptel: cd /usr/src ln -s nameofkerneldirectory linux-2.6 Good luck! t o n y ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
On 6/17/05, David Hajek [EMAIL PROTECTED] wrote: Do you have analog TDM in it? -David Oswaldo Arratia wrote: I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to configure it using [EMAIL PROTECTED] scripts and did not work, so I went the long way and configure with zaptel's instructions and voila! It works like a charm. Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Friday, June 17, 2005 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell PowerEdge + TDM Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David I have 3 sc420's with SATA. All are working well. One has T100P and 1 4 port FXO card. The other 2 have 1 4 port FXO only. t o n y ___ 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] 64 bit
On 5/13/05, Kaj J. Niemi [EMAIL PROTECTED] wrote: How did you get it to compile? Do you have to have a strictly 64 bit compile environment? On RHEL4 it compiles just fine out of the box. Some of the locations are not strictly correct (things get sent to /usr/lib instead of /usr/lib64..) but those are easily fixed when building the rpms. Everything is strictly 64-bit, running mixed 32/64 is just asking for trouble. I also integrated the building of pwlib 1.9.0/openh323 1.17.1 to the whole build process and sound between sip - h.323 users behing Cisco CME systems works great along with using res_config_mysql, cdr_addon_mysql for realtime and cdr logging. :) I spent a few days to figure out the best way of building everything and now usually just drop a cvs snapshot (and/or selective patches from mantis) to stay current. // kaj ___ I am running on SUSE 9.3x64 with very good results. zttest shows 99.9877 (lowest) However the distro supplied the 64bit rpms A.G. (Tony) Nichols I.S. Manager ___ 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] Dlink VPNs??
On Sun, 13 Feb 2005 13:39:09 -0500, Mike Chapman [EMAIL PROTECTED] wrote: Hi, I am thinking of purchasing a cheap Dlink VPN for testing purposes for use with my Asterisk box and would like to ask the list for advice on how to pick a VPN that will work with my box. I am a newbie to both VPN's and Asterisk so any advice will be appreciated. Thanks, Mike ___ I have 2 clients using the 3com secure gateways... never had a problem yet! My cisco 2610 and pix hate it... but some people have luck with them. -- A.G. (Tony) Nichols I.S. Manager ___ 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] PIX Firewall configuration??
On Tue, 1 Feb 2005 08:08:50 -0600 (CST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'd like to open up my firwall so that I can connect my SIP phones to a test server behind or firewall. I can configure an outside addtess to pass traffic to the internal address of the Asterisk server. I'm not sure what other ports need to be opened. My SIP phone will either be at my home behind a linksys cable/dsl router or perhaps at a hotel when I'm on the road. What am I missing in this configuration? Is STUN needed? Pat looks to me that it would be easier to prchase a vpn lic.. I've tried some of the suggestions on the list to no avail. Until version * 1.03 came out... I still had echo/cutouts through the vpn. Now so far knock on wood repeatadly it is going ok. Here is what I found on the list: It works fine for me. I have a handful of Cisco 7960's behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didn't do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context= secret= callerid= host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both 'nat=yes' and 'qualify=yes' so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIX's UDP xlate timeout or the PIX will close the xlate and you won't be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of yes to fine-tine the interval at which it sends a OPTIONS message. Good Luck! t o n y ___ 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] turn on/off auto/attendant by dialing an extension
On Fri, 24 Dec 2004 01:22:47 -0500, Jon Radon [EMAIL PROTECTED] wrote:
Can I ask why? This is clearly the easiest/best way to go about it.
On Thu, 23 Dec 2004 16:21:12 -0500, Tony Nichols [EMAIL PROTECTED] wrote:
The wikki has an example that uses a db
;Login with *801, log out with *802
exten = *801,1,DBPut(auto/attendant=1)
exten = *802,1,DBPut(auto/attendant=0)
;Incoming calls- check if autoattendant is logged in, otherwise goto auto
exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($${autoattendant} = 1?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
Is there a way to do it without the dbput/dbget?
Thanks,
--
A.G. (Tony) Nichols
I.S. Manager
___
--
Is it something someone said, was it something someone said?
I haven't used MySql much... and would like a simple solution I can
copy/paste to other installations.
A.G. (Tony) Nichols
I.S. Manager
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[Asterisk-Users] turn on/off auto/attendant by dialing an extension
The wikki has an example that uses a db
;Login with *801, log out with *802
exten = *801,1,DBPut(auto/attendant=1)
exten = *802,1,DBPut(auto/attendant=0)
;Incoming calls- check if autoattendant is logged in, otherwise goto auto
exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($${autoattendant} = 1?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
Is there a way to do it without the dbput/dbget?
Thanks,
--
A.G. (Tony) Nichols
I.S. Manager
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Re: [Asterisk-Users] IAX hardphone
There was talk on the list... some time ago... an iax firmware has yet to be released. On Wed, 22 Dec 2004 13:35:23 -0500, Dorn Hetzel [EMAIL PROTECTED] wrote: I can't get the link to work. Does this mean that there is some IP phone available which if loaded with the right firmware can do IAX? If so, where can I buy one and where can I get the code? -Dorn On Wed, Dec 22, 2004 at 05:30:48PM +0100, Wilson Pickett wrote: This just in: Centrality has released Version 1.4 for PA168x based phones. Firmwarefiles for different protocols like SIP, IAX etc. can be downloaded from Centrality Website. Firmware for different brands is already available. I've tried out with my HOP-1002 which is actually a WuChuan with PA168S. It connects to my Asterisk and I succeeded to connect to German provider Purtel.com with IAX. Original source: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=HW-phone;action=display;num=1103573331 ___ 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 -- A.G. (Tony) Nichols I.S. Manager ___ 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] Asterisk with SMP hardware
On Wed, 08 Dec 2004 22:32:16 -0600, Andrew Aken [EMAIL PROTECTED] wrote: Does anyone have any experience with running asterisk on multi-processor computers (dual or quad)? Does asterisk on the latest Linux distros take advantage of the extra processors, or does it predominately utilize a single processor? I've been running dual opteron (suse 9.1) for several months now. So far so good! t o n y ___ 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
Re: [Asterisk-Users] IAXy Configuration
On Fri, 2004-11-19 at 15:31 -0800, Erik Espinoza wrote: I can't seem to get this device to grab an ip from dhcp. We have a working dhcp server (unfortunately it is on Windows), but I don't show any leases requested by the iaxy. Anyone have any ideas? The ethernet and phone lines are plugged in before the device is powered. Thanks, Erik I remember a note on the list about issues with a cisco switch, and conecting an iaxy. Mine wouldn't grab an ip either (win2k server), and a cisco 3500. I haven't had time to try a different switch yet. -- Tony Nichols [EMAIL PROTECTED] Appalachian Log Structures Inc. ___ 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
Re: [Asterisk-Users] Can some bady help me ???
On Thu, 2004-11-11 at 20:46, steve szmidt wrote: On Thursday 11 November 2004 04:39 pm, Geoff Nordli wrote: [EMAIL PROTECTED] scribbled on : ok, mathew and other friends I have this package only and I don't now what a have to do with it I repeat im a new linux user I don't now how compile it. I need for start a list of steps to begin or a place where I can get it thanks rodney If Linux is a struggle for you then you may be better of looking at a Live CD type of installation. I haven't tried Xorcom's Rapid installation yet, but it may be worth a try: http://www.xorcom.com/rapid/index.html Geoff Not a bad idea. I just downloaded the SuSe live DVD and it's a very slick system. Easier than anything else out there. I think they also got a Gnome version and KDE version live CD. On my new install of suse 9.1 I do the following 1. during the suse install I pick EVERYTHING (except the united linux box -)! (specific packages are listed on the astrisk website. 2. don't let it do an update (it askes at the end of the installation). 3. after first boot log in; type cd /usr/src/linux (hit enter) type zcat /proc/config.gz .config (hit enter) type make oldconfig (hit enter) Lots of things will scroll by very quickly if you didn't select the kernel source packages you'll get an error here... 4. cd /usr/src (hit enter) then type ln -s /usr/src/linux-2.6.4-54.5 linux-2.6 (hit enter) --- linux-2.6.4-54.5 is the stock 9.1 kernel... if you did an update this will be wrong. 5. cd /where/the folder/that/holds/the/asterisk/libpri/zapte/files I put them in /pbx for simplicity. 6 type cd zaptel (enter) 7. type make clean; make install; cd .. 8. when thats done type cd libpri (enter) hit the arrow up key till you see the same command you typed befor make clean; make install; cd .. 9. when thats done type cd asterisk (enter) then type make clean; make install (enter). 10 that will take several min. --- even with a fast machine. WHen it's done you will see the instructions to type make samples (hit enter). That should get you started! These instructions work with both my intel and opteron based systems. Welcome to the wonderful world of asterisk! t o n y ___ 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
Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)
On Fri, 2004-11-12 at 14:48, JAMES BOTHAM wrote: Hi there, I agree with Greg and also with the documentation group, we are all great at bitching about * (I know I have done a lot of it but thats because UK and support for us is minimal or so it feels) we need to unite, the only reason Microsoft are so popular is because it take 2 minutes to install and applications are usually it is easy to configure (coming from Windows to Suse was quite easy due to YAST but then from Suse to Fedora Core is a nightmare thank god for web min) users a nd administrators don't want to be editing conf files to do the smallest thing i.e. create a dialplan thats a nightmare, coming from an Avaya background (although it has been a year since i touched an Avaya INDeX) you could create powerful and effective dial plans completely graphically it was so easy anybody could do it. Although all system administration was done through a menu driven console and that was really simple to use. We need to take the good from other systems and merge this to Asterisk. Also we have to document it. theres no point in writing the code if nobody can use it. I would like to offer my skills to the production of this I can document, bug test and am great at user interface design I come from a software house background which I can also utilise to get this project off the ground. I suggest that we all meet in a chat room to create some form of a project map and get this off the ground. Cheers James Why not webmin? someone started the interface int (it's in the downloads folder; many admins use it, there are s many plugins for it currently --- so it must not be THAT hard to code. ___ 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] gold rush?
Another new article with asterisk/Digium in mind http://www.onlamp.com/pub/wlg/5909 t o n y ___ 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
Re: [Asterisk-Users] Where to post SuSE 9.x startup script?
I would be interested in the script. Did you do zaptel drivers too? On Wed, 2004-09-08 at 10:41, Martin Mielke wrote: Hi all, I just modified one of the startup scripts provided on the tarball to fit on my SuSE 9.x system to start/stop Asterisk when the system boots or goes down. Maybe I'm overseeing the answer but could't find where to post/(cvs)upload the changes I made... TIA, Martin ___ 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
Re: [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines
On Thu, 2004-08-26 at 09:38, Andrew Kohlsmith wrote: On Wednesday 25 August 2004 23:50, Steve Underwood wrote: Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? I will give this a shot shortly. I still get spandsp segfaulting the odd time so I need to set up a secondary asterisk box to prevent such problems. I've already posted to the list about that particular problem, it's not something as simple as the wrong copy of libtiff or anything. :-) Just FYI; we have a Canon IR3300 fax/copier/scanner big badass unit (over 3mil copies and going...) -A. I too have an ir3300 and was having issues with faxes. I found this googleing: Hooper: IR330 w/ print and fax works just fine. Except that it does not receive from one customer. May I add that it's their most important customer! The faxes are coming from a computer. I have looked at my past note on a similar problem and did this: SSW3 1000-0010 from - SSW17 -0010 from - MEM NL to on from off ATT to 6 NUM 02 to 15 03 to 20 04 to 15 010 to 6800 Not sure of the rom versions. Still isn't receiving. Any suggestions? RussW Try setting SW05 bit 3 to 1 from 0 I have seen this same fault with the HP31xx Hooper: Problem fixed! Changed SSW5 bit three to 1 from zero. I only changed the ssw5 bit the other changes seemed to make it worse. Hope this helps! t o n y ___ 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
RE: [Asterisk-Users] Inter-digit timers on t100
On Wed, 2004-08-18 at 17:01, Kris Boutilier wrote: For the inbound digit problem try adding : debounce=50 ; Needed to reduce the initial off hook debounce in the relevant context for those trunks in /etc/asterisk/zapata.conf Also, are you using 'immediate=yes'? I'll give it a try. Currently I have immediate=no t o n y ___ 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
Re: [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)
I'm running 1 9.1 32bit and 1 9.1 64bit. Both are the shipped kernels and are working very well. t o n y ___ 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
Re: [Asterisk-Users] Inter-digit timers on t100
On Tue, 2004-08-17 at 14:08, Jason Kawakami wrote: Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers or inter-digit dtmf timers or something. I have even tied 2 * together each with t100 cards and have the same problem. Not sure how to proceed, any suggestions? Jason Kawakami I had nearly the same issue between * and an nec pbx. The only way I could get around it was to either make the number a speed dial, or tell the users if the call failed to use the redial button on their phones. However * had issues with the last digits, not the first. The wiki has a section on the inter digit timers, but all the examples I've seen are confined to incoming calls to voicemail. t o n y ___ 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
Re: [Asterisk-Users] IAX2 'no authority found' problem
On Wed, 2004-08-04 at 10:32, Simon Ward wrote:
Hi everyone,
I'm having some problem trying to set up an IAX connection between two *
servers.
The scenario is :
serverA has an X100p card and will direct all calls from the X100p over
IAX to a specific extension on serverB which is at the other end of an
unfirewalled VPN connection.
At the moment serverA tries to redirect the call to serverB but recieves
this message (it appears on both servers) :
-- Executing Dial(Zap/1-1, IAX2/test:[EMAIL PROTECTED]/cardiff) in
new stack
-- Called test:[EMAIL PROTECTED]/cardiff
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 6ms SCall: 1 DCall: 0 [192.168.1.250:4569]
VERSION : 2
CALLED NUMBER : cardiff
LANGUAGE: en
USERNAME: test
FORMAT : 2
CAPABILITY : 65283
ADSICPE : 2
DATE TIME : 151287361
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569]
CAUSE : No authority found
Aug 4 14:50:02 WARNING[147465]: chan_iax2.c:5339 socket_read: Call rejected
by 192.168.1.250: No authority found
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569]
-- Hungup 'IAX2/192.168.1.250:4569/1'
== No one is available to answer at this time
Here are excerpts from the config files :
ServerA:
extensions.conf
[incoming]
exten = s,1,Dial(IAX2/test:[EMAIL PROTECTED]/cardiff)
ServerB:
iax.conf
[cardiff]
type=friend
username=test
secret=test
context=sipfonescard
extensions.conf
[sipfonescard]
exten = cardiff,1,Dial(SIP/1101)
Has anyone got any suggestions on what might be the solution to the 'no
authority found' problem, I'm convinced that it must be something pretty
simple that I'm missing but I can't find any tips to point me in the
right direction.
Any suggestions would be appreciated,
Thanks,
Simon Ward
___
I also have a vpn between my sites.
Here is what I use:
iax (both servers)
[pbx]
type=user
secret=test
trunk=yes
host=dynamic
qualify=yes
username=pbx
servera:extensions.conf
; iax princeton bridge
exten = _2XX,1,Dial IAX2/pbx:[EMAIL PROTECTED]/${EXTEN}
serverb:extensions.conf
; iax ripley bridge
exten = _1XX,1,Dial IAX2/pbx:[EMAIL PROTECTED]/${EXTEN}
Don't use trunk=yes if you don't have a digium card at each pbx (it's
needed for timing).
t o n y
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Re: [Asterisk-Users] Any small colleges/universities using PBX or Voicemail?
On Tue, 2004-08-03 at 08:21, Brian Hudson wrote: What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration path. What equipment was replaced, and how did you handle the loss of investment in any proprietary sets? Many thanks, Brian Hudson Brian, check the last few days of the list - several people have been talking about integrating systems like yours and asterisk. Remember the wiki is your friend! http://www.voip-info.org/wiki-Asterisk+Avaya t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 to IAX2...i'm obviously an idiot!!
On Mon, 2004-07-26 at 14:47, [EMAIL PROTECTED] wrote:
Hi All
I'm trying to get two Asterisk servers to talk to each other using IAX(2).
I've read the WiKi and the docs and tried the examples.
I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960
registering on the other).
I've set them up as follows...
The two servers are set up as friends and have consecutive IP address's.
The setup is that the prefix 3 determines that the server dials the extension number
on the other servers local context:-
extensions.conf
exten = _3,1,Dial(IAX2/OtherServer:[EMAIL PROTECTED]:5036/${EXTEN:[EMAIL
PROTECTED])
The correct port is 4569 for iax2 - the older protocol was 5036
When I do a dial say 32221 this is what comes up in the console:-
Executing GoTo(SIP/2231-, intern-post|32221|1) in new stack
GoTo (intern-post,32221,1)
Executing Dial(SIP/2231-, IAX2/OtherServer:[EMAIL PROTECTED]:5036/[EMAIL
PROTECTED]) in new stack
Called OtherServer:[EMAIL PROTECTED]:5036/[EMAIL PROTECTED]
Warning: chan_iax2.c:1413 attempt_transmit: Max retries exceeded to host
OtherServerIP on IAX2/OtherServerIP:5036/3 (type = 6, subclass = 1, ts=2, seqno=0)
Hungup 'IAX2/OtherServerIP:5036/3'
then the regular cleanup commands
In IAX2 Show Peers I get:-
OtherServerOtherServerIP(S) 255.255.255.255 4569UnMonitored
I'm confused
why is the connection showing on port 4569 in show peers? Is this a default?
Is there a way to test the validity of the IAX2 connection from the console?
Thanks in advance.
P
Here is the iax.conf for both mine:
[pbx]
type=user
secret=test
trunk=yes
host=dynamic
qualify=yes
username=pbx
extensions.conf:
; iax princeton bridge
exten = _2XX,1,Dial IAX2/pbx:[EMAIL PROTECTED]/${EXTEN}
Works for me!
t o n y
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Re: [Asterisk-Users] Echo in asterisk phones.
On Mon, 2004-07-26 at 16:03, albyfromg wrote: Hi all. We have just setup and asterisk with a 4 line zaptel board with Cisco 7960 and BudgeTone-100 IP phones. All works fine except for this nagging echo. Whenever I talk, I hear my voice echo back.. This only happens whenever I talk on an actual phone call. Whenever I talk to another extension, no problems. I would appreciate any help. I have searched the google lists and have not found any relevant info. Thanks. ___ Do you have echocancelwhenbridged=yes in etc/asterisk/zapata.conf ? Also beware if checking it in debug mode (like asterisk -vc) Took me awhile to notice it was going away when I started asterisk normally ban head on wall! t o n y ___ 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
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
On Fri, 2004-07-23 at 11:51, Christopher L. Wade wrote:
Tony Nichols wrote:
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
I used the DTI-U10 (DTI-24-U10). Got it from GTS Telephone
Inc.(732-323-8620) for $300.00 (reconditioned).
My voice t1 comes into asterisk via the first T100P, and attaches to the
nec t1 via the second T100P using em_wink (as a trunk).Then with LCR I
make it add a 9 to the outgoing trunk so asterisk will route it to the
T1.
I grouped the channels that sales calls come into, and I grouped the
channels that go to the nec, so I could use a dial string like:
[sales]
exten = s,1,Playback,transfer|skip ; Please hold while...
exten = s,2,Dial,zap/g7/210 ; Ring, Nec sales group
exten = s,3,Hangup
and to ring extensions on the nec I did this:
; nec bridge
exten = _1XX,1,Dial(zap/g7/${EXTEN})
g7 (group7 is the T1 trunk); Extension 210 is a virtual extension set to
ring 5 other nec extensions; and the 1XX will match extensions in the
100 range that are not on the asterisk.
I should get started on the doc's didn't realize how far I'd come
till now.
Problems I still have:
1. If someone dials slowly from an nec extension - the nec sends the
first group, asterisk then tells them the number is not in service.
2. IAX2 connection to remote office is still choppy occasionally.
Don't know if the pix 501 is getting overwhelmed by the encryption of
voice packets or what?
t o n y
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I haven't really gotten too far into this, but I was wondering just what
'features' of the NEC phones (DTH-16D-1(BK)TEL) I'll be able to work
with from *? I'm currently getting some pressure to ensure that all the
little bells, whistles, and lights will continue to work as they
currently do. I'm also afraid that if the phones would end up becoming
'dumb' (or should I say dumber than they already are), this could kill
my plan for an NEC to * migration, and would be a huge loss for * in my
company, not to mention a huge extra expenditure for ACD Plus otherwise.
Thankfully, regardless of the outcome of everything else, I will be
setting up one machine to interact with some IAXy's for remote extensions.
thanks,
chris
Well your in luck. All the nec features are retained. The only odd-ball
stuff the common user has had to deal with is the line key lights don't
appear like they use to. Originally we had the first 4 marked as local1
-4 then the rest toll1-8. Now when a call comes in button #15 flashes,
and when they pick it up to answer it is dropped on the next available
light (starting at local1). So far it hasn't been a big deal - just
takes some getting use to.
If all else fails you could let the nec answer the calls, and have
asterisk be extensions on the nec . but seems like such a waste.
I hope you have better luck than I with the remote extensions the
iax2 link to my vpn'd offices is a little to choppy... but getting
better.
Good Luck!
t o n y
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Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
I used the DTI-U10 (DTI-24-U10). Got it from GTS Telephone
Inc.(732-323-8620) for $300.00 (reconditioned).
My voice t1 comes into asterisk via the first T100P, and attaches to the
nec t1 via the second T100P using em_wink (as a trunk).Then with LCR I
make it add a 9 to the outgoing trunk so asterisk will route it to the
T1.
I grouped the channels that sales calls come into, and I grouped the
channels that go to the nec, so I could use a dial string like:
[sales]
exten = s,1,Playback,transfer|skip ; Please hold while...
exten = s,2,Dial,zap/g7/210 ; Ring, Nec sales group
exten = s,3,Hangup
and to ring extensions on the nec I did this:
; nec bridge
exten = _1XX,1,Dial(zap/g7/${EXTEN})
g7 (group7 is the T1 trunk); Extension 210 is a virtual extension set to
ring 5 other nec extensions; and the 1XX will match extensions in the
100 range that are not on the asterisk.
I should get started on the doc's didn't realize how far I'd come
till now.
Problems I still have:
1. If someone dials slowly from an nec extension - the nec sends the
first group, asterisk then tells them the number is not in service.
2. IAX2 connection to remote office is still choppy occasionally.
Don't know if the pix 501 is getting overwhelmed by the encryption of
voice packets or what?
t o n y
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Re: [Asterisk-Users] DTMF issue --help
On Fri, 2004-07-16 at 18:45, Andrew Yager wrote:
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digits
and
proceeds; other times it breaks the IVR on the bank side and hangs up.
How do I tell * to stop listning for the DTMF?
; dial a long distance outbound number exten =
_9XXX,1,Dial(${TRUNK}/${EXTEN:1},,Tt)
exten = _9XXX,2,Congestion
Stop telling it to listen to DTMF. It's pretty clear that you just
copied someone's Dial line from somewhere without learning what T and
t do. show application dial on the Asterisk CLI to learn what T and
t do.
If you really need the # transfer, there is a patch on the bug tracker
that implements the use of two keys for transfers (eg ##). I haven't
yet had a chance to test this feature, although I will.
Yours,
Andrew
Thank you Sir,
I'll give that a try I had thought the transfer option would have a
time limit ... not the duration of the call. The boss likes the
secretary to dial the number then transfer to his extension ... so I'll
try that patch -- or maybe make her put them on hold and tell the boss
what line to pickup.
t o n y
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Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, Chris I know what ya mean I've spent nearly $800 in tech time for the Nec guy to help me get mine going. I have the Eletra 192 functioning right now, still have some bugs left but working. I used an Nec T1 card in the electra, and a digium t100p in my * box. Let me know if I can be of any help. When I get the last of the bugs worked out I plan to write down the details and put it on the wikki. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF issue --help
I'm getting down to the last of my * issues ...
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digits and
proceeds; other times it breaks the IVR on the bank side and hangs up.
How do I tell * to stop listning for the DTMF?
Here is my configs:
CVS head 7/14/2004
The call is made Zap to Zap no sip involved.
T100P t-1 (goes to provider ls t1)
T100P t-1 (goes to NEC pbx)
exten listing for outbound ld:
TRUNK=Zap/g2
; dial a long distance outbound number
exten = _9XXX,1,Dial(${TRUNK}/${EXTEN:1},,Tt)
exten = _9XXX,2,Congestion
; Timeout and invalid rules
exten = #,1,Playback(invalid)
exten = #,2,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
Any ideas?
t o n y
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Re: [Asterisk-Users] Config Files
On Thu, 2004-07-01 at 18:27, chouck wrote: Thanks robert, But im having a problem trying to add a user that can login, im using a sipura voip box trying to connect to the server and it always gives me SIP/2.0 403 Forbidden. Under what config can I allow users and hows it work exactly? Thanks again! Try here: http://astguiclient.sourceforge.net/scratch_install.html Matt has a real good walk through installation and even a sample sipura config. t o n y ___ 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
Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse
On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ 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
Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse
Still no go I have asked Digium tech support to look into it. I need the later cvs to get around a bug with the latest tdm400 card (load driver - unload driver - load driver again to make it work. t o n y On Thu, 2004-06-24 at 08:15, Tony Nichols wrote: On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ 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
Re: [Asterisk-Users] Compile Error
On Thu, 2004-06-24 at 13:01, Joseph wrote: Just did a new cvs download and then tried to compile. I get this error message: chan_zap.c:59:2: #error You need newer libpri Then there are some more chan_zap.c errors. Here is the cvs command: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel asterisk libpri And the make command #cd /usr/src/zaptel #make #cd /usr/src/asterisk #make And I did this after moving the current zaptel, asterisk, and libpri to archival. Where do I get this file? Or what am I doing wrong... The correct order is: cd zaptel make clean; make install cd libpri make clean; make install cd asterisk make clean; make install t o n y ___ 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] IAX2 Trunking help!
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched to no avail.
Here is the server a configs (192.168.1.5):
iax.conf
[general]
port=5036
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
tos=lowdelay
register = pbx:[EMAIL PROTECTED]
[pbx]
type=peer
host=dynamic
trunk=yes
secret=test
qualify=yes
extensions.conf
[globals]
TRUNKP=IAX2/pbx:[EMAIL PROTECTED] ; princeton aix trunk
; iax princeton bridge
exten = _2XX,1,Dial(${TRUNKP}/${EXTEN})
Server b config (192.168.2.2):
iax.conf
[general]
port=5036
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
tos=lowdelay
register = pbx:[EMAIL PROTECTED]
[pbx]
type=peer
host=dynamic
trunk=yes
secret=test
qualify=yes
I've tried adding a username=, removing dynamic, adding defaultip=
All have failed.
I'm using cvs head from yesterday. My network is a vpn with a cisco 2600
at site a, and a cisco pix 501 at site b. I have perfect connectivity
between the sites (2 t1's at site a - 1 t1 at site b).
t o n y
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RE: [Asterisk-Users] IAX2 Trunking help!
On Tue, 2004-06-22 at 10:20, David Cook wrote:
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the type=peer/user/friend thing? I did read the docs
but maybe I'm thick. Maybe the visual person in me needs to see a matrix.
Further, If I can get two boxes to talk together like this, what exactly
is the register for ... what does it actually do?
dbc.
Quoting Kevin Walsh [EMAIL PROTECTED]:
David Cook [EMAIL PROTECTED] wrote:
[mycontext]
exten =
_5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _5XXX,2,Hangup exten = _5XXX,102,Hangup
You really don't want your username and password to appear (in
plain
text) in your logs.
Put the sensitive details in iax.conf instead of extensions.conf.
As well as being more secure, it'll make your Dial() string
shorter,
and will mean that you only have to change the connection details
in
one place, should the need arise in the future.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s
h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
If I remove the trunk=yes and the context=local, then change to type
user it seems to work..
I will do some more testing to see if all the different extensions work.
I'm a little bewildered however seems like peer should have been
correct, and trunking=yes.
Guess as long as I can direct calls to/from each location (4 total)
that's all that counts!
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