Re: [asterisk-users] roundrobin and rrmemory with pre-defined agent order
I've also looked into this issue, and it seems that asterisk doesn't respect the order of the members in queues.conf. Asterisk uses a hash table internally to hold the queue members. I guess it's fine when you have dozens of agents, but for simpler scenarios, it's a pain not to be able to determine agent's order. Julian J. M. On Nov 29, 2007 1:46 PM, Fernando Urzedo <[EMAIL PROTECTED]> wrote: > > Hi, > > I would like to implement a queue using a circular strategy, I mean, > using roundrobin or rrmemory strategies. However, I am not able to > define the order Asterisk will call the agents once a new call arrives > in the queue. Seems that Asterisk will always define its order as the > queues.conf file is read, and most of times this order is different from > the one I want (for each queue in queues.conf, I add members in the > order I want them to be called). > > I tried to use the "penalty" setting, but then Asterisk gets stuck in > the first agent (lowest penalty) until it answers a call. > > Is there a way to implement what I am trying? I am using Asterisk > 1.2.19... > > Thanks in advance! -- http://www.julianmenendez.es ___ --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] Free sitting
Freepbx has "devices and users" concept. It may be what you're looking for. You can have your users "log in" in any phone with their extension number and password. After that, all calls to his extension would ring on that phone. http://www.freepbx.org Julian J. M. On 8/6/07, Olivier <[EMAIL PROTECTED]> wrote: > Hello, > > How would you implement free sitting ? > > The idea is to offer teachers the ability to share the same desk and > hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr > Bar is teaching english on wednesdays. > Each has his own extension but use the same hardphone. > > 1. Does a program check a calendar or database somewhere to allocate a phone > to a user (as teachers schedules are known in advance) ? > 2. Every morning, users have to login (logoff is automatic during nighttime) > ? > 3. Users have to login/logoff themselves using a dedicated IVR ? > 4. Users have to login/logoff themselves using a dedicated program on their > PC ? > > Do you offer basic services (emergency and internals calls) between logins ? > Do you use any phone specific menu ? ___ --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] Telco is not detecting HangUp w/ TDM400P
That's ok, and is expected behaviour. The telco will keep the line
open for about 30 seconds. It's useful when there is no PBX, and just
2 or 3 phones attached to the same line... you can hangup on one room,
go to another, pickup and continue the conversation.
Anyway, i guess the telco can reduce that timeout or remove it
completely. Just tell them you have a PBX on that line.
Julian J. M.
On 8/6/07, Alex Pankratov <[EMAIL PROTECTED]> wrote:
> Hi guys,
>
> I spent a couple of hours in Google, but the problem
> appears to be uncommon, so I'd like to ask about it here.
>
> The problem is exactly the opposite to "Asterisk does
> not detect FXO hangup". In my case it's the Telco who
> does not appear to be detecting Asterisk's hangups.
>
> Telco is Telus in Vancouver, Canada. The setup is very
> simple -
>
> Telco -> FXO/TDM400p -> * -> softphone
>
> The log is -
>
> -- Starting simple switch on 'Zap/4-1'
> -- Executing [EMAIL PROTECTED]:1] Answer("Zap/4-1", "") in new stack
> -- Executing [EMAIL PROTECTED]:2] Dial("Zap/4-1", "IAX2/alex|5|r") in new
> stack
> -- Called alex
> -- Call accepted by 192.168.1.102 (format gsm)
> -- Format for call is gsm
> -- IAX2/alex-2 is ringing
> -- Nobody picked up in 5000 ms
> -- Hungup 'IAX2/alex-2'
> -- Executing [EMAIL PROTECTED]:3] Hangup("Zap/4-1", "") in new stack
> -- Hungup 'Zap/4-1'
>
> At this point the caller (say, me on my cell phone) still
> sits connected and enjoying the white noise. The longest I
> waited was about 20 seconds and then I hung up.
>
> Similar problem is described here (November 2006) -
>
> http://lists.digium.com/pipermail/asterisk-dev/2006-November/024768.html
>
> but there's no solution and the discussion is not very
> helpful.
>
> Any pointers and/or ideas are greatly appreciated.
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Re: [asterisk-users] Dropouts and echo
What kind of switch are you connecting the phones to? I've seen that behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it with a different one fixed the problem. Julian J. M. On 7/31/07, Tom Lanyon <[EMAIL PROTECTED]> wrote: > The issues: > Dropouts - by far the most serious issue we've encountered. On most > calls (normally anything longer than 1 or 2 minutes), suddenly one > end of the call will go silent and not be able to hear the other > person. After a few seconds of "I can't hear you!" the audio returns > and continues normally. This seems to happen whether it's an internal > call between SIP devices or whether it involves a call via our ISDN > gateway. At first we believed this was just when we had our phones on > 'speakerphone' and that it was an issue with the physical SIP phone > itself, but we're now also finding 'dropouts' just using the phone > handset aswell. ___ --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] Re: wrong values in duration and billsec in CDR
On 3/23/07, C F <[EMAIL PROTECTED]> wrote: On 3/22/07, Tomislav Parcina <[EMAIL PROTECTED]> wrote: > C F wrote: > >> So, how to solve this problem? > > > > Get an ISDN line, or maybe just VoIP. > > This really isn't answer to my question ;) Why not? FXO is answered as soon as you go off hook. There is no real way it will work on FXO, unless you get an ISDN or all VoIP lines. Actually some telcos use polarity reversals to signal answer and hangup states. That's what answeronpolarityswitch and hanguponpolarityswitch parameters in zapata.conf. Julian J. M. ___ --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] ChanSpy and MeetMe
You are using parameter b in ChanSpy arguments. That will only select
unbridged channels, Zap/73 is connected directly to the meetme
application. Remove that 'b' and try again.
Julián J. M.
On 3/22/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I have been successful using ChanSpy on a standard Dial call but when
attempting to ChanSpy on an incoming call that has been added to a MeetMe
conference (attempting to coach an agent that is speaking to a conference of
callers) it seems to fail to connect to the channel. Here's the console
dump:
-- Accepting call from '2154182700' to '3399' on channel 0/18, span 4
-- Executing [EMAIL PROTECTED]:1] Answer("Zap/90-1", "") in new stack
-- Executing [EMAIL PROTECTED]:2] Read("Zap/90-1",
"GOTDTMF|demo-instruct|1||1|1") in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'demo-instruct' (language 'en')
-- User entered '5'
-- Executing [EMAIL PROTECTED]:3] GotoIf("Zap/90-1", "5?9") in new
stack
-- Goto (from-internal,3399,9)
-- Executing [EMAIL PROTECTED]:9] AGI("Zap/90-1", "simpleconf.agi") in
new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/simpleconf.agi
-- Playing 'digits/5' (language 'en')
-- AGI Script Executing Application: (CHANSPY) Options: (Zap/73|wbq)
I verified Zap/73 is the correct channel of the caller currently in the
conference I am attempting to ChanSpy on. Has anyone done this before? I
apologize in advance if my question lacks the necessary information, I'm new
to Asterisk.
-George
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Re: [asterisk-users] Rebooting ALL polycom phones
for i in `seq 100 150` ; do asterisk -rx "sip notify polycom-check-cfg $i" ; done Julian. On 3/12/07, Mike <[EMAIL PROTECTED]> wrote: Hi, I know that if you have Polycom phones properly configured, you can use "sip notify polycom-check-cfg SIP_REGISTRATION_ID" to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike ___ --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] Give "Busy" to the 3rd call on a BRI using chan_capi
I don't use chan_capi, but bristuff. http://www.junghanns.net/en/download.html Julian. On 2/3/07, Armin Schindler <[EMAIL PROTECTED]> wrote: On Sat, 3 Feb 2007, Julian J. M. wrote: > I'm still using asterisk 1.0.x bristuffed at one site.. Is there > anything similar for this? When both channels are in use, 3rd call > doesn't recive busy signal, but a message fromt he TelCo (something > like "The dialed number is not currently available"). The asterisk version has nothing to do with that. Which chan-capi do you use? Armin > On 2/3/07, Armin Schindler <[EMAIL PROTECTED]> wrote: > > What type of line is that? The 'base number' is also a MSN on lines I > > know. > > Or is it PtP with DID? > > > > Armin > > > > On Sat, 3 Feb 2007, Cosmin Prund wrote: > > > > > Thanks, it really was easy. > > > Unfortunately it only works for "MSN's", not for the "base number". > > > Oh well, > > > I'll just stop using the base number, I've got enough MSN's anyway. > > > > > > Thanks again. > > > > > > Armin Schindler wrote: > > > > On Thu, 1 Feb 2007, Cosmin Prund wrote: > > > > > > > > > Any ideas? It should be simple... > > > > > > > > > > > > > It is easy: read the README in chan-capi.org package ;-) > > > > > > > > Just look into the variable BCHANNELINFO and you will know if > > > > it is a > > > > call > > > > without b-channel (the third call). > > > > > > > > Armin > > > > > > > > > ___ > > --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
Re: [asterisk-users] Give "Busy" to the 3rd call on a BRI using chan_capi
I'm still using asterisk 1.0.x bristuffed at one site.. Is there anything similar for this? When both channels are in use, 3rd call doesn't recive busy signal, but a message fromt he TelCo (something like "The dialed number is not currently available"). Thanks, Julián J. M. On 2/3/07, Armin Schindler <[EMAIL PROTECTED]> wrote: What type of line is that? The 'base number' is also a MSN on lines I know. Or is it PtP with DID? Armin On Sat, 3 Feb 2007, Cosmin Prund wrote: > Thanks, it really was easy. > Unfortunately it only works for "MSN's", not for the "base number". Oh well, > I'll just stop using the base number, I've got enough MSN's anyway. > > Thanks again. > > Armin Schindler wrote: > > On Thu, 1 Feb 2007, Cosmin Prund wrote: > > > > > Any ideas? It should be simple... > > > > > > > It is easy: read the README in chan-capi.org package ;-) > > > > Just look into the variable BCHANNELINFO and you will know if it is a > > call > > without b-channel (the third call). > > > > Armin > > > ___ --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] Voice Recognition
"My voice is my passport; verify me." ;) I don't think you'll get reliable results with 8khz sample rates. The highest frequency wave you can achieve is a 4khz square wave. Anyway, i don't think if such software exists ;) Julian J. M. On 1/19/07, Asterisk <[EMAIL PROTECTED]> wrote: Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support "voice recognition" (not "speech recognition") - task of recognizing people from their voices? Thanks, Alex ___ --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] Re: Codec swap (reinvite)
While I was trying to patch chan_sip.c to force a specific codec by
using a channel variable, I found out that this is already
implemented. It there even for asterisk 1.2
sip.conf:
allow=g729,gsm,ulaw
outbound call:
exten => _X.,1,Set(SIP_CODEC=ulaw)
exten => _X.,2,Dial(SIP/itsp/${EXTEN})
inbound call:
[from-pstn]
exten => _X,1,Set(SIP_CODEC=ulaw)
exten => _X,2,Answer()
Julian J. Menendez
On 10/15/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
On 2006-10-14 20:00:30 -0700, "Julian J. M." <[EMAIL PROTECTED]> said:
> I've finally given up on trying to fax over my Digium TDM400 card.
> I've found that fax over VoIP is quite more reliable (at least I can
> receive the faxes).
>
> My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes
> everyday (just ocasionally), i pretend on using g729, unless a fax is
> detected.
>
> Is there any way to force asterisk to make a reinvite, and swap the
> codec on the fly? Something like this would be great:
>
> exten => fax,1,CodecChange(ulaw)
> exten => fax,2,rxfax(blablabla)
I think the answer is no. I am pretty sure this has been discussed
multiple times and there is currently no way to change the codec once
the call is established.
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Re: [asterisk-users] Detect IP path before calling
Use qualify=3000 For an acceptable lag of up to 3 seconds. That value _doesn't_ mean to ping the peer every 3 seconds, btw. By default, It will be pinged every 60s if ok, and every 10s if there is any problem (peer lagged, unreachable, etc). Julian. On 1/4/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: qualify=yes in sip.conf in each [whatever] section on sip.conf should track if the far end is at least responding to SIP messages. My problem is that if the far end device takes too long to respond to a SIP OPTIONS packet, Asterisk will consider it lagged. I've not found any of making qualify'd devices be considered reachable %100 of the time when there is no actual problem. If you are using SIP VoIP providers and failover to another route, qualify=yes might be something to try. If you need reliable qualify's you might consider using the money you would spend on writing a monitoring script and use it to pay a bounty to add "qualify smoothing" to SIP similar to that feature in IAX2. ___ --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] vzaphfc?
Remove from zapata.conf the lines re bristuff (bri_cpe_ptmp, etc).
Setup misdn:
/etc/init.d/misdn-init config
vi /etc/misdn-init.conf(check it's ok, NT or TE, PTP or PTMP...)
/etc/init.d/misdn-init start
chkconfig --add misdn-init
Setup chan_misdn, in /etc/asterisk/misdn.conf. At the end:
[telco]
port=1
context=from-pstn
msns=*
Then, in extensions.conf:
exten => _,1,Set(CALLERID(num)=00)
exten => _.,2,Dial(misdn/g:telco/${EXTEN})
Julian J. M.
On 1/2/07, Remco Barendse <[EMAIL PROTECTED]> wrote:
On Fri, 29 Dec 2006, Julian J. M. wrote:
> It's not necessary to recompile the kernel for mISDN support. Check
> http://www.laimbock.com/asterisk/
>
> Grab the mISDN source rpm, and build it.
>
> $ wget
>
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm
> $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm
>
> then check /usr/src/redhat/RPMS/i386/
> You should have the kernel modules and userspace applications. Once
> installed, I could enable chan_misdn in asterisk 1.4 without issue,
> and it's working great in NT mode with ISDN phones. I haven't tested
> asterisk 1.2, but there is no it shouldn't work as well.
I deleted all the bristuff modules i could find plus the old asterisk
libs, compiled zaptel, libpri and asterisk from scratch but can't get it
to work.
First I get errors about something I guess is missing from misdn, later
errors about zaptel.
I'll just toss the HFC-S card and convert the ISDN line to analog.
These are the errors :
.mISDN_close: fid(14) isize(131072) inbuf(0x2a96ee6010) irp(0x2a96ee6010)
iend(0x2a96ee6010)
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: "ports=(null)" (section: intern) invalid or out of range.
Please edit your misdn.conf and then do a "misdn reload".
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: "ports=(null)" (section: first_extern) invalid or out of
range. Please edit your misdn.conf and then do a "misdn reload".
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: "ports=(null)" (section: second_extern) invalid or out of
range. Please edit your misdn.conf and then do a "misdn reload".
P[ 0] Got: 1 from get_ports
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method 'bri_cpe_ptmp'
Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must
be specified before any channels are.
Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so:
load_module failed, returning -1
Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module
chan_zap.so failed!
Cheers!
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Re: [asterisk-users] vzaphfc?
It's not necessary to recompile the kernel for mISDN support. Check http://www.laimbock.com/asterisk/ Grab the mISDN source rpm, and build it. $ wget http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm then check /usr/src/redhat/RPMS/i386/ You should have the kernel modules and userspace applications. Once installed, I could enable chan_misdn in asterisk 1.4 without issue, and it's working great in NT mode with ISDN phones. I haven't tested asterisk 1.2, but there is no it shouldn't work as well. Julian J. M. On 12/29/06, Remco Barendse <[EMAIL PROTECTED]> wrote: On Thu, 28 Dec 2006, Gavin Hamill wrote: > On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: > >> vzaphfc is not a complete replacement of bristuff. It replies on most of >> it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI >> driver for HFC-s-based PCI cards. > > Further, if you're looking for 'something else' re: cheapo ISDN cards, > definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, > no wacky stuff.. all Asterisk-core support that worked really well in the > brief time I tested it. > > The key difference is rather than generating 8000 interrupts per second, the > mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) > polls the card, leading to much lower system load, and no 'wanted 8 bytes, > read 7!' errors from dmesg. Thanks for the tip, I'll have a look at it. The main reason for me to use bristuff is that i don't want to mess mess around downloading and compiling my own kernels. I am just running CentOS 4 boxes with stock CentOS 4 kernels. Everytime I was screwing around with making my own kernels sooner or later I got bitten by screwing up the installation of the kernel and the box wouldn't boot anymore. :) On the wiki I found the manual from BeroNet which looks pretty straightforward but is for Asterisk 1.2 Any differences for Asterisk 1.4? Thanks!! ___ --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] agi+cepstral driving me nuts
Why don't you try app_swift?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
This one even compiles on 1.4, and has buffering, meaning that it
doesn't have to wait for the tts to generate the complete output.
http://www.loopfree.net/app_swift/
exten => s,1,AGI(getinfo.php)
exten => s,2,Swift( ${RESULT_INFORMATION} )
Julián J. M.
On 12/26/06, blackwater dev <[EMAIL PROTECTED]> wrote:
I just got cepstal working fine in the dial plan using code like:
exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
enter your zip code.)
The php script it calls is based on the nerdvittles weather one so it calls
a webpage which prints to the screen, the nerdvittles code uses system to
generate the .wav file then has the dial plan call it via:
//php script
$retcode2 = system ("flite -f $tmptext -o $tmpwave") ;
//extensions
exten => 411,9,NoOp(Wave file: ${TMPWAVE})
exten => 411,10,Playback(${TMPWAVE})
Since I am using capstral, I simply changed the line to below which works
fine from the command line but when calling, I never hear it, it just hangs
up. Is it timing out? Is there a better way to do this? How can I return
just a string of Text to read so I don't have to create the .wav file then
play it?
$retcode2 = system ("swift -n Diane -m text -f $tmptext -o $tmpwave") ;
Thanks!
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[asterisk-users] SIP/IAX Fax Detect on Asterisk 1.4
Hello, Has anyone managed to compile app_nvfaxdetect on asterisk 1.4? Is there any other way of detecting incoming fax calls on non-Zap channels? Julian. ___ --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] qualify=yes
FYI, the interval at which the device is checked is 60seconds when OK, and 10s when not OK. It can be changed in channels/chan_sip.c. Look for this lines: #define DEFAULT_FREQ_OK 60 * 1000 /* How often to check for the host to be up */ #define DEFAULT_FREQ_NOTOK 10 * 1000 /* How often to check, if the host is down... */ If the device (hard or softphone) doesn't support keepalives and the nat router has a short timeout (less than 60s), even when qualify=yes, the nat mapping will timeout, thus being unable to receive calls. In this case, you can lower that 60 to a value slightly lower than the router timeout. Julian J. M. On 11/22/06, Pavel Jezek <[EMAIL PROTECTED]> wrote: qualify=xxx in sip means, consider peer as OK if delay reply is bellow xxx (ms) qualify checks (POKE) is every 60s (and is not configurable in sip.conf) qualify setting in iax.conf is working differently, this is how frequently to check peer (and is not possible to set some POKE delay threshlold as working qualify in sip) this is quite misleading and inconsistent and should be improved ;-) PJ Vicky wrote: > I doubt that . I think qualify=500 means asterisk checks every 500 ms > if the > other extension is available or not . Because when qualify=( value in > ms ) > is set and you do a sip show peers in console asterisk whos how much > latency > is there between extension and asterisk . If i set qualify = no then it > shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows > extension lagged if latency is less than 10 ms ... It just checks > every 10 > ms for extension . I am not very sure though :) > > On 22/11/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: >> >> Enrico Pasqualotto wrote: >> > Enrico Pasqualotto wrote: >> >> hi all, how can I set the interval in second from retrasmit the magic >> >> packets when qualify is set to on? >> > You have to set qualify=second instead of qualify=yes|no. >> >> This is WRONG. qualify=500 means "consider this device lagged if >> responses take longer than 500ms" I don't know if you can set the >> frequency of qualify packets. If you can, I assume the option would be >> listed in sip.conf.sample. >> ___ >> --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
Re: [asterisk-users] Very high translation costs for g729
Try forcing asterisk recalculate those costs: CLI> show translation recalc 20 Julian J. M. On 11/5/06, Avi Miller <[EMAIL PROTECTED]> wrote: Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI> show translation Server 1: g729 -26252525252426 -5336 Server 2: g729 -66656565656469 -9075 On my other three boxes, I get much saner vaules (costs anywhere from 3 to 6). Any ideas why two boxes have such high costs? All the servers run the same OS, updated to the same versions of everything, including kernel. Four of the five boxes run x86_64 kernels, with the two that are playing up both running x86_64 kernels. I've switched the entire network to using Speex instead of g729 until I find out why I'm getting such high numbers here. I suspect (but can't prove) that this may have been the cause of some audio issues between these two servers as the phones on either end use alaw, so Asterisk is transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . > > Open Source - Own It - Squiz.net .. /> ___ --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] blind transfers with IP Polycom 501
Yes, digitmap... If you just want to allow any digit pattern, use this digitmap: xx.T x -> Any valid digit . -> 0 or more ocurences of previous charracter T -> Default timeout (3 seconds) Any digit followed by a 3 second timeout will match. You can include pattern to match * and #. xx.T|*x.T|#x.T Julian. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: > > > > > I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only > problem I'm experiencing is the following: I can't seem to get blind > transfers to work with my Polycom 501 phones Either through the feature > code or the soft keys. > > > > > > Feature code blind transfers: > > I set up a feature map in features.conf like this: > > blindxfer => # > > This works for all my softphones, just not the 501 phones. > > > > Soft key Blind Transfers: > > Then I tried blind transfers through the phone like this: > > Transfer à Blind Key à Extension. > > Here's the problem. If I enter a 2 digit extension, it works. > > Example: Transfer à Blind Key à 70 (this parks my calls without a problem) > > If I try to blind transfer an extension with three or more digits, the phone > cancels the blind transfer. > > Is there something obvious I'm missing here?? > > > > Thanks in advance. > > > > > > > > > ___ > --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] Codec swap (reinvite)
Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on the fly? Something like this would be great: exten => fax,1,CodecChange(ulaw) exten => fax,2,rxfax(blablabla) Thanks, Julián J. M. ___ --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] Help - call recording being cut short if transferred
Add /n to you Local dial string, i.e.: Dial(Local/1234/n) From http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels : Adding "/n" at the end of the string will make the Local channel not do a native transfer (the "n" stands for "n"o release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. Julian J. M. On 8/5/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: Using svn trunk, I am trying to record a call coming in from a zap line. This works: A) Zap->StartRecording->DialSip->xferToSIP2->Talk->Hangup The entire call is recorded until the hangup However, if you use a local channel instead: B) Zap->StartRecording->DialLocal->DialSip->xferToSIP2->Talk->Hangup the conversation is recorded up to the xferToSip2 part (i.e. the conversation between Zap and SIP2 is not recorded) If I add /n to the local channel C) Zap->StartRecording->DialLocal/n->DialSip->xferToSIP2->Talk->Hangup The entire call is recorded until the hangup, BUT the call quality is extremely poor, and I get a whole heap of warnings on the console (see below). Why does B) not work ? I would have thought that recording the zap channel would continue *until* the zap channel hung up, regardless of what goes on with the other channels. Any light on this would be most appreciated. I've got my *ss kicked for this (we do a lot of transfers) and I've been told to "sort it" for this weekend ;( Julian [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) [Aug 5 09:07:28] WARNING[26717]: chan_sip.c:3342 sip_write: Asked to transmit frame type 8, while native formats is 4 (read/write = 8/4) ___ --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] app background
Have you tried "CLI> show application background" ? exten => s,1,Background(myfile|n) Julian. On 7/31/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Hello friends, I want to use the background(playfile) application without the channel being answered. I dont want playback because I would like the callee to dial the number while the file is being played. but I dont know how do i do that. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
I didn't test it with a Sipura, but a TDM400. You can check this page for configuration codes for the F251M. http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In Spanish). If the SPA-3000 supports detecting polarity reversals, you'll need them. Julian. On 7/26/06, Jon Farmer <[EMAIL PROTECTED]> wrote: Hi I have been asked if it possible to connect a SE F250M to Asterisk. I have never used one of these devices before but from what I have gathered they need a FXO interface. As the Asterisk box is hosted remotely would it possible to use a Sipura 3000 to provide the FXO interface and successfully use the F250M. If anyone has any pointers on this I would be grateful. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] NAT and externip problem or bug
Why don't you use the syntax that I mentioned in my first reply?
According to http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+localnet
The correct syntax is:
localnet=192.168.0.0/255.255.255.0
Keyword localmask is deprecated in asterisk 1.2... And btw, you should
have seen it in the logs. According to chan_sip.c, around line 12508:
} else if (!strcasecmp(v->name, "localmask")) {
ast_log(LOG_WARNING, "Use of localmask is no
long supported -- use localnet with mask syntax\n");
}
Julian J. M.
On 7/22/06, Robert Jenkins <[EMAIL PROTECTED]> wrote:
The simple thing is that if I have 'externip' set, I can see on a soft phone
(running on a PC on the same local subnet as asterisk) that it's seeing a
call from another local device as coming from [EMAIL PROTECTED] - which is
the external IP and as everything is inside the firewall there is no audio
from the soft phone when the call answered.
If I comment out the 'externip' line & restart asterisk, the soft phone then
correctly sees the local call as being from [EMAIL PROTECTED] and I get
two-way speech.
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Re: [asterisk-users] NAT and externip problem or bug
Have you made sure you are also setting localnet in sip.conf? externip=1.2.3.4 localnet=192.168.0.0/255.255.255.255 Asterisk won't use externip for devices on your local network. Julian. On 7/22/06, Robert Jenkins <[EMAIL PROTECTED]> wrote: Hi, I've recently got asterisk running on it's own pc inside my firwewall. Mostly it's working fine, but there is one silly problem I can't figure out. (For reference, Asterisk is the latest stable version as of last weekend 14th July. All connectivity is SIP or IAX). I initially had 'externip' set to my public IP. I have the appropriate 5000 range ports forwarded to the asterisk PC and external calls seem OK. The 'local' phones are a mixture of Sipura boxes and softphones. Problem: No or one-way audio in internal calls. Reason: Asterisk appears to be using the 'externip' address for all SIP devices, regardless of their NAT setting. Once a call starts, some softphones change the address they are responding to & use the external IP rather than the asterisk PCs local IP on the same subnet... I have tried all NAT options and spent quite a while reading everything I can find about sip.conf, but I can't so far find any way of changing this behaviour. All the internal phones work fine if I comment out the externip line, but then the connections outside the firewall are likely to have problems. Is there any way of configuring externip on a per-device basis, or should it only have effect on NATed devices? Thanks, Robert Jenkins. ___ --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] ISDN: 3° incoming call
BRI ISDN is 2 channels, what would you want to do with a 3rd call? Julian On 6/30/06, francesco giuliani <[EMAIL PROTECTED]> wrote: Does any boby knows how to manage a 3° incoming call in a BRI ISDN line by chan_modem? ___ --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] asterisk-stat display problems
Check /var/log/http/error.log Usually, asterisk-stat fails because it tries to use more memory than allowed in php.ini. Julian J. M. On 6/26/06, Chris Earle (CBL) <[EMAIL PROTECTED]> wrote: yep I don't know exactly which things the php-gd is used for, but like I said, someof the pages work, like the main record page, the little red bars showing call volume work fine Really annoying, cause it looks so good at that point, then you go to use the other pages/features and it's broken Thanks for the reply, -- Chris - Original Message - From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, June 26, 2006 1:49 PM Subject: Re: [Asterisk-Users] asterisk-stat display problems > do you have the php-gd package installed on your * server? > > Chris Earle (CBL) wrote: > > Hey all, > > > > having a terrible time with asterisk-stat -- it runs, server is fine, but > > some of the pages don't display properly/at all --- I think this is a code > > problem with them, but not sure. I thought everyone loved the asterisk-stat > > package? > > > > See below problems. Any ideas? Areski hasn't replied to me since > > > > -- > > Chris > > > > > > - Original Message - > > From: "Chris Earle (CBL)" > > To: "Areski" > > Sent: Tuesday, June 13, 2006 6:15 PM > > Subject: Re: CDR-Analyser version question > > > > > >> Thank you for the reply; > >> > >> I see now that the main file cdr.php does work with argument ?s=1, 2, > >> etc > >> but when s=0, does not load > >> > >> I get an Apache error : > >> > >> relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: > >> gdFontCacheShutdown > >> > >> Not sure if that means anything important; > >> > >> > >> > >> > >> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the > >> pages do not complete their output -- no search button displayed, stops > >> outputting radio buttons for UserField row etc > >> > >> So at this point, only the main Call-log page (s=1) works. > >> > >> > >> I am using Debian with php 4.4.1 > >> Mysql ver 12.22, Distrib 4.0.24 > >> GD Library is 2.0.33 I think > >> > >> > >> Any input you can pass along would be much appreciated! I am comfortable > >> with php so if you want me to modify sourcecode that is fine > >> > >> Thanks! > >> > >> > >> > >> > >> - Original Message - > >> From: "Areski" > >> To: "Chris Earle (CBL)" > >> Sent: Sunday, May 28, 2006 7:11 PM > >> Subject: Re: CDR-Analyser version question > >> > >> > >>> No there is no asterisk requirement to make asterisk-stat. > >>> Indeed the soft is only based on the cdr database. If you have an error > >>> you can give me more info, I may help you. > >>> > >>> Rgds, Areski > >>> > >>> On 5/25/06, Chris Earle (CBL) wrote: > >>>> Hi there, > >>>> > >>>> quick question: > >>>> > >>>> Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk > >> 1.0.x > >>>> and can't get it to load the cdr.php properly > >>>> > >>>> so I downgraded to v1.3 and it works... > >>>> > >>>> Let me know if there's an asterisk version requirement for each > > version > >> of > >>>> the CDR Analyser > >>>> > >>>> Thanks! > >>>> > >>>> > >>>> > >>>> -- > >>>> Chris Earle > >>>> > >>>> > > > > > > > > -- > Mojo <[EMAIL PROTECTED]> > Office Manger, Horan & Company, LLC > (907) 747- x112 > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Asterisk and SATA Raid 1
Hi, I also remember reading that.. but i'm not sure if it was Digium's word ;) It had to do with some SCSI and SATA controllers taking control of the PCI bus for too much time, and causing frame-slips or IRQ losses on TDM hardware. Julian. On 6/5/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: - mustardman29 <[EMAIL PROTECTED]> wrote: > I know that Digium and FreePBX were not recommending it awhile back > but I > think that was based on 2.4 Kernel and Digium hardware issues. I am Can you give me a pointer to any place where Digium recommended against using hardware RAID cards? I can't imagine that being an issue for Asterisk or any of our hardware. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] SPA-941/942 Bulk provisioning
Hi, You can have a look here http://blog.julianmenendez.es/sipura It's drupal based provisioning system for linksys and sipura phones. You'll need to register an account to use it. Basically, you have profiles (linksys na-pap2, sipura spa-3000, etc). You choose one to create a base "configuration". After that, you create one or more "devices" based on that configuration, which inherit its settings. A device is identified by its mac address. For real 0-config provisioning, one would just need a dhcp server, and a tft server with an init.cfg file like this: http://blog.julianmenendez.es/sipura/device/xml/$MA 30 30 Yes Julian. On 5/3/06, Ed <[EMAIL PROTECTED]> wrote: [EMAIL PROTECTED] wrote: > I'm in the process of writing an autoprovisioner which can handle > fresh out-of-the-box linksys, snom, and grandstream with 0-config > (other than entering the mac into a textfile). You never have to touch > the phone. Just plug it in. any result? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed
Try adding 'r' to the dial options. According to "show application dial": r- Indicate ringing to the calling party. Pass no audio to the calling party until the called channel has answered. exten => 3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50, r) Julian. On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]> wrote: > > > > Hello Everyone. I usually find my own solutions for problems but this time, > after several months, I've given up. > > > > My asterisk is set up so that incoming calls from my voip provider ring on > both my sip extension and my cellphone at the same time. When the system > receives an incoming call, ringtones indicating that the call is being > connected play normally for the first 5 seconds to the caller, but they > suddenly stop as the call to my cellphone starts to make progress. This > causes some people to hang up, despite the fact that the call is still being > connected. Callers who stay on the line are able to talk to me on either > the sip extension or the cellphone once I pick up either one. > > > > I have tried a lot of workarounds like including a priority to answer the > incoming call, invoke the playtones command before the dial command, but > this doesn't seem to work either. Can anyone replicate the problem? Have I > ran into a bug? I have pasted as much info as I deemed relevant; please let > me know if I'm missing something. Thanks. ___ --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] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
I have 2 different instalations with 1 Billion HFC Card (1port), and 1 TDM400. Asterisk 1.0.10+bristuff+florz patch. Only issue is that you must load all modules (wcfxs, zaphfc) before runing ztcfg, otherwise nothing works. Everything works ok, even faxing. Julian. On 3/30/06, Chris Earle <[EMAIL PROTECTED]> wrote: > What? After hours of searching for anything to help me, I found this > comment about zaptel cards in systems with bristuff-cards (junghanns for me > in this case) > > I havent' seen any other reports of this sort of behaviour --- can anyone > confirm whether they've got a QuadBRI and TDM400P card working together in > one machine? > > > thanks :-S > > > > "Zoa" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > > > > >We stopped with the bristuff as bristuff will break any other zaptel > >cards in the same system. (pri seems logical, why the tdm card also > >broke is unknown to me). > > > ___ > > --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 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] H323 behind a Firewall
The h323 channels doesn't have any support for NAT. You'd need to register with a properly configured gnugk for that. Julian J. M. On 3/29/06, Alberto Sagredo <[EMAIL PROTECTED]> wrote: > If you open h323 port and rtp ports, it should work. > > Il Neofita escribió: > > There is a proble to put an H323 Asterisk server behind an iptables > > firewall? > > > > > > > > > > > > ___ > > --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] SMS in Spain (it seems Protocol 2)
Are both protocols enabled? I remember I had to first send an SMS with the Domo (an analog phone with sms capabilities) before I could even receive them. Maybe protocol 1, even if it's implemented, needs to be enabled someway. Julian J. M. On 3/29/06, Fran <[EMAIL PROTECTED]> wrote: > Telefónica use both protocols to deliver an SMS (UBS1 and UBS2). > Most nowadays fixed-devices (in Spain) are UBS2 but there are UBS1 too. > The Telefónica messaging platform have the information of terminals of each > subscriber and its access protocol. > > good luck, hope it helps!! > Fran > > > > -Mensaje original- > De: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] nombre de Alberto > Sagredo > Enviado el: miércoles, 29 de marzo de 2006 17:19 > Para: Asterisk Users Mailing List - Non-Commercial Discussion; > [EMAIL PROTECTED] > Asunto: Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2) > > > Capatres released some time ago a solution with an ITSP. > > Maybe it could help > > http://blogs.capatres.com/index.php?op=ViewArticle&articleId=18&blogId=1 > > Carles Pina i Estany escribió: > > Hello, > > > > > > (I have asked it some time ago in Asterisk-es mailing list, so excuse me > if > > anybody receive it twice.) > > > > I am trying to send SMS in Spain using landlines. It seems that > > app_sms.c only handles Protocol 1, but Spain and Italy are using > > Protocol 2. > > > > I have been searching in Internet without any results... anybody is > > sending SMS from Asterisk (or any method) using Protocol 2? (so, it > > seems, Spain or Italy?) > > > > If nobody is able to send, is there more people interested on it? Or any > > project/person/firm trying to send SMS using Protocol 2? > > > > Thank you very much, > > > > PD: some guy from Asterisk-es said to me that it seems that Telefonica > > wants to implement Protocol 1 too... but I don't have any information > > about deadlines, etc... > > > > > > > -- > Alberto Sagredo > Departamento Técnico > Peoplecall > > > Email : [EMAIL PROTECTED] > Blog: http://www.voipnovatos.es > > Tel./Ph. : +34 91 120 5080 > Tel. Dir./Dir. Ph.: 700 757 139 > Fax./Fax.: +34 91 661 9460 > > > ___ > --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] sipura spa2 + asterisk bug ?
That ATA cannot do 2 simultaneous calls with g729. The second call is probably trying to use ulaw, alaw or g723. Are you sure any of them are enabled for that extension? Julian. On 3/27/06, Tofik Suleymanov <[EMAIL PROTECTED]> wrote: > Hello, > > How to reproduce this bug (?) : > > 1. register sipura spa2 with 2 lines on asterisk. > 2. use first line to call somewhere. > 3. while using first line try to call from second line somewhere else > > in 3 step i hear fast busy tones on second line and asterisk console > gives me this short error: > > Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No > compatible codecs! > > My sipura adapter is using g729a codec. > When using both of sipura lines separately everything works fine, > until someone tries to use both lines simultaneously. > > Any advice ? > > > Tofik Suleymanov > ___ > --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] asterisk as a fax server
For converting email to fax, you have asterfax (http://asterfax.sf.net) For fax2email, app_rxfax is well documented. Check http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk You can also use hylafax (with iaxmodem or chan_fax). It may give you finer control of incoming faxes. Julian. On 3/23/06, nik600 <[EMAIL PROTECTED]> wrote: > hi > > is it possible to build a fax server with asterisk? > > i would like to make a system that: > > - receives email, converts email and attachments as image and send it via fax > - receives fax, converts fax as an image an send it attached in a > email to a specific address > > obviusly the asterisk server is configured with a ISDN card on the > phone network... ___ --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] Action after _caller_ has hungup(cmd Dial 'g'-option)
You can use DeadAGI.
exten => _X.,1,DeadAGI(agicall.agi,${EXTEN})
now in that AGI (pseudocode)
$exten=Get parameter 1
$dialstring="SIP/mytrunk/".$exten;
$res=$agi->dial($dialstring),
//If we used deadagi, if the _caller_ hangs up, the agi keep runing here
$chres = $agi->channel_status();
$status=$chres['data'];
Here's a list of possible return values. If $status==6, then the
_callee_ hung up.
CLI> show agi channel status
Usage: CHANNEL STATUS []
Returns the status of the specified channel.
If no channel name is given the returns the status of the
current channel. Return values:
0 Channel is down and available
1 Channel is down, but reserved
2 Channel is off hook
3 Digits (or equivalent) have been dialed
4 Line is ringing
5 Remote end is ringing
6 Line is up
7 Line is busy
---
Julian J. M.
On 3/10/06, Christian B <[EMAIL PROTECTED]> wrote:
> Hello!
>
> There's the "g"-option for the Dial-cmd that allows to execute the next
> extensions in the current context when the callee hangs up.
>
> I would need the same for a call where the caller hangs up, concretely
> i have to inform a agi-application of the end of a call. Does someone
> know a way to do this from the dialplan?
>
> thanks
> Christian
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Re: [Asterisk-Users] Hangup issues
Hello, Load wctdm with debug=1, i.e, add this line to /etc/modprobe.conf: options wctdm debug=1 Then watch /var/log/messages (tail -f /var/log/messages will do it), and check when you are getting the first polarity reversal, you should get it before the first RING. If it happens that you get it when asterisk answers, that would explain your problem. BTW, is it a pstn line? or a gsm fct? If the later, you need to set it up for proper hangup detection in asterisk. Julian J. M. On 3/7/06, Carlos Prieto <[EMAIL PROTECTED]> wrote: > Hi ! > > I have some issues, i don't know exactly if it's a busy detection issue. > > When i dial into the Asterisk box, and if i hang up before the Asterisk > answers with the IVR Welcome message, the Asterisk goes on with the call. > But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk > hangs up too. > > I have this parameters on zapata.conf: > > busydetect=no > answeronpolarityswitch=yes > hanguponpolarityswitch=yes > callprogress=no > > > I've tested with different values por busydetect set t yes and several > busycount values. > > > I'm using Asterisk 1.2.4 and Zaptel 1.2.3 with a Digium TDM400P with 2 FXO > modules and Kewl Start Signalling. > > Thanks in advance for the help. > > ___ > --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] How to route incoming calls to different contexts?
what about this?
[incoming]
exten => DID1,1,Goto(incoming1,${EXTEN},1)
exten => DID2,1,Goto(incoming2,${EXTEN},1)
Julian.
On 3/5/06, Tele Cost Price Reducer <[EMAIL PROTECTED]> wrote:
>
> hi Zach,
> i would use GOTOIF to forward the DID from within the [incoming] context to
> the other context. i would try :
> exten => gotoif($[did]=DID1,goto did1|s|1,)
> exten => gotoif($[did]=DID2,goto did2|s|1,)
>
>
>
>
>
> On 3/4/06, Zach A <[EMAIL PROTECTED]> wrote:
> > Both DIDs are SIP and from the same provider. Format of registration is
> > like this:
> >
> > sip.conf
> >
> > [general]
> > bindaddr=xxx.xxx.xxx.xxx
> > port=5060
> > context=incoming
> > disallow=all
> > allow=g726
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> > dtmfmode=rfc2833
> > canreinvite=no ; required for incoming calls to ring extensions
> > insecure=invite ; outgoing call not working without this
> > tos=0x18
> > nat=yes
> >
> > register=DID1:[EMAIL PROTECTED]
> > register= DID2:[EMAIL PROTECTED]
> >
> > [DID1]
> > username=DID1
> > type=peer
> > secret=1234
> > host=xxx.xxx.xxx.xxx
> > fromuser=DID1
> >
> > [DID2]
> > username=DID2
> > type=peer
> > secret=1234
> > host=xxx.xxx.xxx.xxx
> > fromuser=DID2
> >
> > Now both DIDs are sent to context [incoming] which is the default
> > context for SIP. If I add context=incoming2 under any DID section, it
> > doesn't go to that context and still go to the default context. How can
> > I direct DID2 to [incoming2] context?
> >
> > Zach A
> >
> >
> > -Original Message-
> > From: Joseph Tanner [mailto:[EMAIL PROTECTED]
> > Sent: Friday, March 03, 2006 7:18 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] How to route incoming calls to
> > differentcontexts?
> >
> > First, tell us if it's sip, iax, or zap. Then tell us what provider
> > (most will use the same general config, but some like ipkall are
> > special and a bit tricky).
> >
> > joseph Tanner
> >
> > On 3/3/06, Zach A <[EMAIL PROTECTED]> wrote:
> > >
> > >
> > >
> > > Hi everybody,
> > >
> > >
> > >
> > > It should be a simple thing to do but I don't know how to do it. Now I
> > have
> > > 2 DIDs and I want one of them go to [context1] and other one to go to
> > > [context2]. How can I achieve this.
> > >
> > >
> > >
> > > Thanks,
> > >
> > >
> > >
> > > Zach A
> > > ___
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> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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>
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>
>
>
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Re: [Asterisk-Users] Two FXOs getting bridged?
You don't seem to have disconnect supervision enabled. Julian. On 3/2/06, Warren Burstein <[EMAIL PROTECTED]> wrote: [...] > One additional mystery is that I don't know why these calls persist. > When I hang up either of the bridged extension on my test system, the > bridged call ends. When a single outside call is hung up on the other > side, asterisk notices. I don't have enough phone lines and cellphones > to test if this works when two outside lines are bridged. Does external > hangup detection work on your system? ___ --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] IAXModem/Hylafax problem
If you can read Spanish, check http://blog.julianmenendez.es/asterisk-hylafax-iaxmodem Julian. On 2/23/06, Bob McDowell <[EMAIL PROTECTED]> wrote: > > I think I'm very close to getting IAXModem and Hylafax going, but my > current inbound hylafax logs show this: > > Feb 23 10:09:37.98: [ 3638]: MODEM > Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier > > Two questions - > > 1) Does anyone know what step I missed here? (I.e. please help!) > 2) Is there a document I should be working off of? Google doesn't seem > to think so... > > Bob McDowell > > > > ___ > --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] Grandstream GXP-2000 Auto Answer
Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom >From that article: There is an 'allpage.agi' now available at http://aussievoip.com.au/allpage.agi. Documentation is available in the file. This should work with Snom and Grandstream GXP2000 phones (and possibly budgettones if they roll the changes across) with firmware greater than 1.0.13 (not publically available at time of writing, due out in October 2005) I've used that with my GXP-2000, and seems to work ok. I had, however, to adapt it to my needs. Regards Julian J. M. On 12/17/05, William M. Sandiford <[EMAIL PROTECTED]> wrote: > > Has anyone been successful getting Auto-Answer by Call-Info to work with the > GXP 2000 > > I have followed the suggestions in > > http://www.voip-info.org/wiki/view/GXP-2000 > > Specifically I have: > > 1. Upgraded to 1.0.1.13, which supposedly supports this feature > 2. Set Allow Auto-Answer by Call-Info to YES in the GXP2000 config > 3. Used, SIPAddHeader(Call Info: answer-after=0) in my dialplan prior to > the Dial command. > > Still the phone just rings, and doesn't auto-answer. > > Any suggestions? > > Thanks in advance, > Bill > ___ > --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] Why Won't Asterisk REINVITE?
Try removing the Answer() before the Dial... e.g.:
[spa2100]
exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
exten => _X.,2,Dial(SIP/netvoice-102)
exten => _X.,3,Hangup
Regards
Julian J. M.
On 12/9/05, George Pajari <[EMAIL PROTECTED]> wrote:
> Eric "ManxPower" Wieling wrote:
>
> > T/t/H/h and other options to Dial require Asterisk to stay in the RTP
> > stream.
>
> Understood but already checked as not being the cause. Thanks for the
> suggestion, though.
>
> Here is our entire extensions.conf context:
>
> [spa2100]
>
> exten => _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
> exten => _X.,2,Answer
> exten => _X.,3,Wait(2)
> exten => _X.,4,Dial(SIP/netvoice-102)
> exten => _X.,5,Hangup
>
> where
>
> [netvoice-102]
> accountcode=netvoice-102
> callerid=NETVOICE COMMS <604 484 8647>
> username=netvoice-102
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> nat=no
> qualify=no
> mailbox=102
> context = netvoice-internal
> canreinvite=yes
> disallow=all
> allow=ulaw
>
> Here is a "sip show channels" during a call:
>
> aa.bb.cc.39netvoice-1 7f6a484c36f 00103/0 ulaw
> aa.bb.cc.40nvc.test.a 6cfe5077-2f 00103/00102 ulaw
>
> --
> George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
> Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
> www.netvoice.ca www.ip-centrex.ca
> www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
>
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Re: [Asterisk-Users] Zaptel + No Hangup
No... It applies without problems (just a little offset) Julian. On 10/27/05, Giovanni Miano <[EMAIL PROTECTED]> wrote: > Any problems with bristuff ? > > 2005/10/26, Julian J. M. <[EMAIL PROTECTED]>: > > You can try this patch > > (www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your > > telco sends your polarity reversals on answer and hangup. > > > > Julian J. M. > > > > On 10/26/05, Giovanni Miano <[EMAIL PROTECTED]> wrote: > > > I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2 > > > > > > All works perfectly but command "Hangup or Hangup()" in dialplan dont > > > hangup call > > > > > > (zapata.conf within busycount=4 and busydetect=yes) > > > > > > Why ? > > > > > > Country is ITALY > > > > > > -- > > > Giovanni Miano > > > ___ > > > --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 > > > > > -- > Giovanni Miano > ___ > --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
Re: [Asterisk-Users] Zaptel + No Hangup
You can try this patch (www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your telco sends your polarity reversals on answer and hangup. Julian J. M. On 10/26/05, Giovanni Miano <[EMAIL PROTECTED]> wrote: > I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2 > > All works perfectly but command "Hangup or Hangup()" in dialplan dont > hangup call > > (zapata.conf within busycount=4 and busydetect=yes) > > Why ? > > Country is ITALY > > -- > Giovanni Miano > ___ > --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
Re: [Asterisk-Users] iaxmodem
For hylafax to answer a call, you need to use faxgetty.. Add this 2 lines to your /etc/inittab and run "init q" to force a reload: IAX:2345:respawn:/usr/local/bin/iaxmodem ttyIAX modem:2345:respawn:/usr/sbin/faxgetty ttyIAX Change the paths according to your system. Julian J. M. On 10/25/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Probably you are right > > I installed hylafax and configured it to use iaxmodem, but I didn't start > it > Now I will research how to start hylafax, and I will try again > > Andrea ___ --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] Does anyone run Asterisk on FC4? with Digium's TDM40B cards
Run memtest86 from the boot menu. You may have faulty RAM. I had the same problem installing CentOs 4... Julian J. M. On 8/6/05, Kumara Jayaweera <[EMAIL PROTECTED]> wrote: > Hi all, > Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success > stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). > Please any comments? > > Kumara ___ 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] zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
Try Florz patch with your bristuffed asterisk. Better support for missed interrupts. Julian J. M. On 7/22/05, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: > Hi, > I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs > modules and everything seemed allright but linux log shows the following > message: > zaphfc: sync lost, pci performance too low. you might have some cpu > throtteling enabled. > Anybody knows what it means? ___ 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] VPN's
Make sure, you include remote office's lan in the localnet directive (otherwise, they'll use the wan ip address, and that may be the problem...) Julian. On 7/15/05, Peter Osborne <[EMAIL PROTECTED]> wrote: > Hi All, > > I'm using Asterisk for my PBX, I have a remote office that is connected by a > VPN link. I am using Openswan on my side and a Linksys box on the remote > side. I have a Polycom IP300 on the remote side configured with a static IP > address. When I call the phone on the remote side, it rings and establishes > the call fine. The problem I am having is that the remote side can hear the > call find but the local side hears nothing. Because of the VPN there are no > firwalls in the way. Does anyone have some ideas or atleast how I can track > down the problem. ___ 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] * behind NAT and local subnet
Have you set correctly the externip and localnet keywords in sip.conf? Julian. On 7/15/05, Damon Estep <[EMAIL PROTECTED]> wrote: > I have an * box behind a NAT router (static NAT, port ACLs set up correctly) > > Most of the SIP users are on the local subnet with the * box, they work fine > > Take one of the same users off of the local subnet and come in through the > NAT router and these results; ___ 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] Support needed
Have you tried googling for "asterisk e164" ? Julian. On 7/13/05, Will Velez <[EMAIL PROTECTED]> wrote: > Hi my name is Will Velez. > Does Asterisk support E164? > Thanks ___ 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] SNOM 360 and parking
To do attended transfers with Snom 360, you need to put the current call on hold, dial the dest extension, tell him/her something, and press the Transfer button. I don't think it'll work with asterisk call parking, though... Julian J. M. On 7/12/05, Patrick Friedel <[EMAIL PROTECTED]> wrote: > OK, last showstopper that I just can't puzzle my way through - parking > calls with the snom phones. I get the two phones connected, I hit > transfer on one, the other phone goes to MOH and the first phone gives > me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM > hangs up before I have a chance to hear which extension it parked to. > Is there a way to make the SNOM phones stay off hook until you > explicitly hang up during a transfer? (my only complaint about these > phones - occasionally they're just too darn smart for their own good.) > > I can live without actual snom-style orbits at this time (handy though > they might be), since the current system involves parking the call on an > external line and walking over to another office to say that they have a > call. I imagine that down the road it'll usually just be an attended > transfer, but we do park calls around phones a fair bit as we brainstorm > issues. > > (Actually, I can't get attended transfers working, either. All > transfers are blind. Related?) ___ 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] Incoming calls from BudgetPhone.nl
Try using insecure=very in your peer definition. That makes asterisk not require authentication from your peer if comes from the ip address give in host=xxx.xxx.xxx.xx directive. That helped me receiving calls from my sip provider, which had exactly the same problem. Julian. On 7/10/05, Peter Raaijmaker <[EMAIL PROTECTED]> wrote: > (this time with subject) > > Hello, > > I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. > When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy > tone. > I tried X-lite, which worked perfect, so my modem (with nat) probably is not > the problem. > I did a sip debug and got the following output. > Because I'm new to Asterisk I can't get the error why this is not working. > To me it all looks fine, no warnings or what so ever… > > The settings in sip.conf and extensions.conf are identical to those of > http://www.voip-info.org/tiki-index.php?page=Talkin2ya > > Does anyone know what I'm doing wrong > > Thanks, > Peter. > ___ 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] URGENT: hardware spesifications needed
I guess the wrong word in the original mail was URGENT... Julian ;) On 7/7/05, Michael L Smith <[EMAIL PROTECTED]> wrote: > Who are you to decide what Information can and cannot be "legitimately be > sought here:? ___ 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] FXO hangup Problem.....
Where are you located? What's not working is the remote party hangup detection, and callprogress only works on selected countries. Please, load your wcfxs (or wctdm) module with debug=1, and check /var/log/messages to see if the card is detecting polarity reversals when you answer the PSTN line and when the other party hangs up. If that's true, you may want to try this patch: http://www.maxosystem.net/asterisk/asterisk-stable-polarity.html Julian. On 7/7/05, Nahid Hossain <[EMAIL PROTECTED]> wrote: > Hello, > I am getting problem for delay call hang-up with the below scenario: ___ 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] Help needed - Zap Transfer Failing...
Why don't you just use Dial(SIP/125)?? Or better, if you have your extensions defined in context e.g. [from-internal], just do: exten => 9876,1,Goto(from-internal,125,1) Julian. On 7/8/05, Mark Edwards <[EMAIL PROTECTED]> wrote: > Hi. > > I have the following line in the default context of all my internal > extensions: > > exten => 9876,1,Transfer(125) > > When I dial extension 9876 from any sip phone, * dutifully transferrs it to > extension 125, which is just what I want. > > Unfortunately when I dial 9786 from my Zap connected analogue phone, the > transfer doesn't go through and the dialplan drops through to a hangup. ___ 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] Early media dectection problem
It may be a problem with your sip phone, as some doesn't support early media connect, and you just hear local ringback until the call is answered. I had exactly this kind of problem until Swissvoice (IP10s) released last firmware. Snom has no problems neither. Julian J. M. On 7/5/05, kurt x <[EMAIL PROTECTED]> wrote: > I noticed when I call certain IVR systems, such as 1800calldhl, that > Asterisk will not > barge the prompt. Would this imply that Asterisk has an Early media > detection problem. > Is anyone else experiencing this problem. Is there a fix? ___ 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] Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Recheck your zaptel.conf. That's not the correct setup for a T1 trunk. You need to know the signalling the channel bank uses, and specify the voice channels (bchannel=1-24), and the signalling channel (dchannel=25). Those numbers are bogus, as I've never worked with T1 ;) BTW, why are you using such setup (1 channel bank to connect to 24 analog lines) instead of asking your Telco to install a T1 trunk in your office? Julian. On 7/5/05, Mehran Mozaffari <[EMAIL PROTECTED]> wrote: > Hi, > > I have some problem to get this setup working. I have a CAC Channel > Banl I, with FXO and an Asterisk box ( I am using [EMAIL PROTECTED] 1.2) > and I have a TE110p installed in this box. > > What I want to do is, just to be able to dial one of those lines that > already are connected to the channel bank, and transfer that call > through TE110p and Asterisk to a user agent somewhere through > Internet. > > Agent> > > At this time the SIP UAs can communicate with each other and > everything works properly, but I can't dial through channel bank. When > I dial one of those numbers, I will get no answer ring, and I can't > see anything coming to Asterisk through CLI. and when I tried to dial > through SIP UA to the PSTN end, I will get all circuits are busy now > from asterisk. > > Any Idea what should I do? at this time all lights are green and it > looks like that everything is working properly, but I am not sure > where is the problem, here are my settings: > > /etc/zaptel.conf: > > span=1,1,0,esf,b8zs > fxols=1-24 ___ 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 box after an analogic pbx
exten => _X.,1,Dial(Zap/1/0www${EXTEN})
That doesn't wait for dialtone, just dial 0, sleep for 1,5sec, and
dial the number.
Julian.
On 7/5/05, Accursio Avona <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I'm newbe with asterisk and i'm facing with this problem that i'm not
> able to solve.
> I've to put an asterisk box after an analogic pbx wich require a 0 digit
> to give the dialtone.
> So when a client ask asterisk to dial an extension it should
>
> 1) send the 0 digit
> 2) wait for the dialtone
> 3) dial the extension the client send.
>
> How can i obtain this result?
>
> Thank's in advance
> Best regards
> Accursio Avona.
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Re: [Asterisk-Users] Calls authentication by IP address
You can try insecure=very for your peer (in sip.conf). Make sure, they don't have to register -> host=123.123.123.123 instead of host=dynamic. Julian. On 7/5/05, VoIP Newbie <[EMAIL PROTECTED]> wrote: > Hi all, > > Is there any AGI supported calls authenticated by IP address? > > Many thanks. > Newbie > ___ > 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
Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
I had a similar problem with a recent install: * TDM11B * 1 Port HFC Card I got that messages about HDLC Framing errors. It ended up being the way I loaded the required kernel modules. 1) Remove or comment "install" lines in /etc/modprobe.conf (or modules.conf), regarding the kernel modules for the TDM and the HFC Card. They usually launch ztcfg. 2) In the init script, load both modules manually (modprobe wcfxs zaphfc) 3) Issue the ztcfg command 4) Load asterisk That way it worked without problems. Julian J. M. On 6/27/05, David Masure <[EMAIL PROTECTED]> wrote: > > > Hi all, > > I'm using asterisk 1.0.6 with bristuff-0.2.0-rc7k. I've already set up 3 > boxes with the same config, but I'm facing something strange with the fourth > one : > > In my messages log, I've got thoses lines : > > 0xff). > kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 13, stat = > 0xff). > kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 11, stat = > 0xff). > kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = > 0xff). > > This problem causes asterisk to crash... The only thing to do is a nice > reboot :-) > > I read some informations about it in the list telling that's because of a > wrong signalling type... But as, I am using the same config everywhere, I > don't believe the problem is on the box ... > > Questions > > 1) The asterisk box is located in France...so could this be that the telco > (France Telecom) is powering it's lines with different signalling (for > example mutli point and point to point ?) > > 2) Does anyone ever face the same problem ? > > 3) Does someone know how to cope with this problem ? > > Thanks for your help ...I'm really don't know what to do !!! > ___ 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] GXP2000 and hint LED's
I've just checked the download page, and the latest firmware available is 1.0.1.8. Where did you find 1.0.1.9? This phone has some nasty bugs, one of them being that the other end HEARS you after you press the Transfer button and you hear a dialtone. It doesn't send any message to asterisk so that it can play music on hold to the caller. Julian. On 6/9/05, James Bean <[EMAIL PROTECTED]> wrote: > Asterisk 1.0.7 > > Has anyone got the hint function working, and maybe with the GXP2000. > > I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment > trying to get the LED's to light up. ___ 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] 180 Ringing? (BUG?)
I guess that's "Early Media Connect", i.e., if the phone supports that (not all do), the channels get bridged just after dial completed, (SIP 183), and what you hear is the remote ring tones (from your telco), not locally generated (as if it received SIP 180 Ringing). What IP phones are you using? You may try xlite and check if you hear ringing tones. Julian. On 6/9/05, Mirko Marghitola <[EMAIL PROTECTED]> wrote: > Voilà. > Now i know where is the problem. > I use 2 ISDN channels with a with a fritz! card and the junghanns capi > drivers. > The problem appears with SIP to ISDN calls. > > The SIP 180 ringing message doesn't appear because the ISDN PBX sends > the "ALERT" message in-band (channel B), and not in the D channel. So > Asterisk doesn't know when the ISDN channel is ringing. > With my configuration Asterisk can not understand the in-band signalling > for the capi channels, is it possible to use "in-band" signallisation > for capi channels? > > Mirko ___ 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] TDM400P... ignoring hanguponpolarityswitch
I've made a backport of this patch for asterisk stable. You can get it here: http://www.maxosystem.net/asterisk . The page is in Spanish, but you just need to download and apply the patch to chan_zap.c. It also works with bristuff patch applied. Julian J. M. On 6/9/05, Neil and Fiona <[EMAIL PROTECTED]> wrote: > Is there a list of options that are valid for stable? I downgraded from > Head to stable when I had IAX trunking problems (one way audio) with a > VSP. So I am using my conf files from Head, which could be the problem. > > I've got a copy of sample config files from 1.07 (Or I think they are, I > didn't label it well when I archived it). It seems to have the option in > it. > > There has been a patch in Head for the IAX2 trunking problem, so I think > I could go back to head. ___ 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] TDM400P... ignoring hanguponpolarityswitch
I've used that feature in asterisk HEAD, and it has worked for me (i needed to apply a little patch for it to work for incoming calls also), but i also used answeronpolarityswitch=yes. Maybe it's a logic bug in the code. Try with that option and tell us the results ;) BTW, it doesn't matter is the module detects the idle polarity as 1 or -1... The code only checks for the polarity switch event (-1>1 or 1>-1) Julian. On 6/8/05, Neil and Fiona <[EMAIL PROTECTED]> wrote: > I've just had polarity reversal provisioned by our telco to test hangup > detect with a TDM400P > > I've set hanguponpolarityswitch=yes in zapata.conf > > When I start Asterisk I get "ignoring hanguponpolarityswitch" > in /var/log/asterisk/messages > > I assume that the option is either not valid or conflicts with another > setting somewhere. > > Any ideas? ___ 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] How to handle one incoming call on multiple lines?
Isn't it easier to talk to your Telco, and tell them to just ring the first free line, instead of all 4? Julian J. M. On 6/8/05, Erwin Lubbers <[EMAIL PROTECTED]> wrote: > Hi, > > I have connected 4 analog public telephone lines to an Asterisk server using a > Digium TDM400P card and that working fine. But my 4 lines are connected to > each other in a group by the telecom operator. So if someone calls me all 4 > lines are ringing. I wrote a AGI script which will handle the incoming calls, > but before it decides to answer the call or not the next channel is ringing > and the script is started again. How can I create a situation that after the > first ringing channel is coupled to a script the other channels are still > ringing for the same call? ___ 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] Double NAT issues with SIP and workaround (?)
Hello, I've been fighting one-way-audio issues with asterisk and SIP extensions for some time..., and I want to share with you my findings ;) My setup: * 1 ADSL router (Zyxel) * 1 Asterisk box with private IP, and interesting ports forwarded to it. * Several extensions, some local some remote The problem: * External extensions behind double nat don't get audio when they initiate a call. But if the extension receives the call, there is no problem. The fix: * Get a Linksys WRT54G, and setup the adsl router in bridge mode, giving the public IP address to the WRT. * Setup port forwarding and QoS (optional) * Enjoy VoIP ;) I don't know the exact reasons why this happens, but it works ;). Julian. ___ 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] secretary function
Try this: 1) You're on a call 2) Push a Line button, so that you get dialtone 3) Dial the boss extension # 4) Hey boss, you have a call from XXX 5) Push Transfer 6) You can select which call to transfer (if you have more that 1 on hold) 7) Push transfer again. Julian. On 6/3/05, Christian Hiller <[EMAIL PROTECTED]> wrote: > Hello, > > we got a SNOM 360 here and this gota TRANSFER button. > With this i can transfer a call from my phone another one. But when i > push this Button and transfer the call to another phone, i get kicked out. > > Now, every secretary first asks the chief if he is available or not - > how can i implement this feature ___ 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] Incoming calls picked-up then simply hanged-up
Are you sure you have context=from-pstn in your zapata.conf for the fxo channels? Julian. On 5/12/05, fhunter <[EMAIL PROTECTED]> wrote: > I don't think my first posting went thru. > > I am trying to set up Asterisk for the first time. I am new to this. > I am using [EMAIL PROTECTED] > I have a TDM400P with one FXO and one FXS > > The system is working for outgoing calls and if I test incoming calls using > . > But when doing an actual call the system seems to answer the call and then > immediately hang up. > The incoming calls are set up to go from the PSTN to the Digital > Receptionist. > But I get the same behavior if I have incoming call send to the extension I > have set up. ___ 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] Log Output
In /etc/asterisk/logger.conf, add this: full => notice,warning,error,debug,verbose Then watch /var/log/asterisk/full getting really big ;) Julian. On 5/11/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Guys. > > Is there a way to output the same information shown on the console when > invoked as - but to a log file for later grepping and such? > > I noticed the normal log only shows warning and errors but no info messages > like in the console.. Any ideas? ___ 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] Ericsson FCT f251m and polarity reversal
Hello,
This is a little off-topic. I have an Ericsson FCT f251m, according to
the specs it supports call signalling through polarity reversals and
loop break, but it's currently disabled.
On my PSTN line, my TelCo does send polarity switchs to signal answer
and hangup (answeronpolarityswitch=yes and hanguponpolarityswitch=yes
in zapata.conf), and asterisk detects it alright.
I've been unable to find an admin manual for this fct (or any of the
250 series). Can someone point me to the manual or just give me the
activation code to dial?
Thanks in advance,
Julian J. M.
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Re: [Asterisk-Users] Re: HINT
But that only works when SIP/201 receives a call, right?
What if SIP/201 is making a dialout call, does it show as busy in the
phone's keypad?
Julian J. M.
On 5/7/05, Thorben Jensen <[EMAIL PROTECTED]> wrote:
> > Could you please give us some more detail as to what you did, in terms of
> > configuring the hint, and specifically what changes in the behavior of the
> > running server-phone interaction as a result?
>
> You need to set the hint for the phone when the phone is being dialed like
> this:
>
> exten => 201,hint,SIP/201
> exten => 201,1,macro(dial-sip,201)
>
> It's important that you write the full name of the phone "SIP/201" as you
> can't use substitutions like this SIP/${EXTEN} - it took me a long time to
> figure that out.
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Re: [Asterisk-Users] Put a wait in a .call file.
Add some 'w' before the number, i.e., Zap/g0/ww1812121212 Julian J. M. On 5/4/05, Ronan Eckelberry <[EMAIL PROTECTED]> wrote: > Does anyone know of a way to put a wait or a pause in a .call file? > When my * tries to make an outgoing call on a Zap channel, it does not > wait for a dialtone. It just starts dialing. ___ 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] TDM users: modified zttest.c for testing
I also had problem faxing with spandsp with my old server (Athlon 700 on a VIA chipset). Now I've instaled asterisk on a P4 2.8Ghz (Asus P5P800, btw great board, let's you assign the preferred interrupt for each PCI slot), with 256Mb, and here's what I get (unpatched zttest): (before I never got to 100%) [EMAIL PROTECTED] zaptel]# ./zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% --- Results after 19 passes --- Best: 100.00 -- Worst: 99.987793 I have yet to try spandsp, but I think i'll work without problems. Julian J. M. ___ 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] b0rked hfc config
Shouldn't it be: ? bchannel => 9,10 dchannel => 11 bchannel => 12-13 dchannel => 14 Julian J. M. On 4/27/05, Thomas Andrews <[EMAIL PROTECTED]> wrote: > On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote: > > > I have 2 Billion cards and I can't get the hfc driver to work. I get > > this error: > > > > ZT_CHANCONFIG failed on channel 9: No such device or address (6) > > > > What am I doing wrong ? > > This is my /etc/asterisk/zaptel.conf: > > [channels] > > switchtype = euroisdn > signalling = bri_cpe_ptmp > pridialplan = dynamic > prilocaldialplan = local > nationalprefix = 0 > internationalprefix = 00 > echocancel=yes > echotraining = 100 > echocancelwhenbridged=yes > immediate=yes > group = 1 > context = incoming > channel => 9 > channel => 10 > channel => 12 > channel => 13 > > ___ > 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
Re: [Asterisk-Users] Alternatives to SpanDSP??
Hello Colin, Did setting the latency timer really helped? What latency do you set for the rest of pci devices? just 0? Julian J. M. On 4/26/05, Colin Anderson <[EMAIL PROTECTED]> wrote: > 2. ZTTEST is a critical metric. I was getting disconnects on about 20% of > faxes until I looked at the output of ZTTEST and found that it was dropping > below 99.98% occasionally. Using setpci I changed the latency on the Zaptel > boards (T100P & TDM04) to the max, 254 and cranked down the latency on > everything else as low as I dared. Now, I get 99.9873% across the board as > long as I run the test, and I even get the magic 100% on 1 in 10 test > passes. ___ 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] Alternatives to SpanDSP??
Hi Steve, I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting <200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: cat /proc/interrupts CPU0 0: 23411526 XT-PIC timer 2: 0 XT-PIC cascade 4: 80 XT-PIC serial 8: 1 XT-PIC rtc 10: 23322936 XT-PIC wctdm 12: 1 XT-PIC acpi 14: 91663 XT-PIC ide0 15: 51573 XT-PIC eth0 NMI: 0 ERR: 0 $ ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% Thanks Julian J. M. On 4/26/05, Steve Underwood <[EMAIL PROTECTED]> wrote: > Why would you expect a bunch of fax modems to work any better than > spandsp? If spandsp doesn't work reliably your system is very likely broken. > > I have had hundreds of complaints about spandsp reliability. I have > analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which > has real spandsp problems. The rest had frame slips. Of the 5 or 6 with > real problems, most have been fixed in the latest version. I have one > weird audio log from a new HP combination printer and fax machine that i > haven't sorted out yet. These HP machines really are total crap. I have > workarounds in spandsp for several blatently wrong things they do. I > don't yet know who is at fault with this latest problem. > > Regards, > Steve ___ 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] Trouble with call parking/transfer
Make sure you have canreinvite=no in your sip peers definition, and/or that you pass 't' or 'T', to the Dial statement. Julian J. M. On 4/25/05, Tim Pushor <[EMAIL PROTECTED]> wrote: > Hi all, > > I am still unable to initiate a call transfer with the keypresses > defined in features.conf in a couple month old version of asterisk from > CVS HEAD. ___ 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] can't make my PRI dial out
I haven't worked with PRI, but could it be related to an invalid callerid?
What about:
exten => _X., 1, SetCallerId(123123123)
exten => _X., 2, Dial(Zap/g1/${EXTEN})
Julian.
On 4/22/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> On April 22, 2005 11:48 am, Mark Phillips wrote:
> > Nothing happens. I get the same (non)error.
> > I get plenty of output when receiving a call however.
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[Asterisk-Users] Fax and spandsp
Hello, I've read every message on this list regarding faxing and spandsp/rxfax, so this is a desperate email seeking for help... maybe someone can tell me what's going wrong. I have a working TDM400 with 1FXO and 1FXS port, 1 fax machine connected to the FXS port. I can send and receive faxes without problems, using asterisk as a bridge between the fax machine and the PSTN. No frame slips nor lost interrupts I guess. I want asterisk to receive incoming faxes (via rxfax application) and send them by mail. The problem is that, although the fax machine and the asterisk log report a succesful transfer, the tiff file is just 5-8kb. I've tried several libtiff packages, always recompiling spandsp and asterisk after installing, but always the same problem. My system: Athlon 700Mhz, VIA chipset. USB/Sound/2nd IDE/parport disabled, just the basics. Fedora Core 3: libtiff-devel-3.6.1-9.fc3 libtiff-3.6.1-9.fc3 spandsp-devel-0.0.2-1 spandsp-0.0.2-1(compiled from the 0.0.2pre15 tarball) Runing in text mode For testing, I've setup a special extension, which is called by the fax machine. BTW i've also tried using an external modem with hylafax, connected to the FXS port and got the same results. Here in this log, you see the image size, just 287 rows, 37 bad rows, image size 0 bytes :-? CPU usage didn't go higher than 5%, according to top. Apr 19 21:05:14 DEBUG[7768]: == Apr 19 21:05:14 DEBUG[7768]: Pages transferred: 1 Apr 19 21:05:14 DEBUG[7768]: Image size: 1728 x 287 Apr 19 21:05:14 DEBUG[7768]: Image resolution7700 x 7700 Apr 19 21:05:14 DEBUG[7768]: Transfer Rate: 9600 Apr 19 21:05:14 DEBUG[7768]: Bad rows37 Apr 19 21:05:14 DEBUG[7768]: Longest bad row run 21 Apr 19 21:05:14 DEBUG[7768]: Compression type2 Apr 19 21:05:14 DEBUG[7768]: Image size (bytes) 0 Apr 19 21:05:14 DEBUG[7768]: == Apr 19 21:05:23 DEBUG[7768]: == Apr 19 21:05:23 DEBUG[7768]: Fax successfully received. Apr 19 21:05:23 DEBUG[7768]: Remote station id: TestingId Apr 19 21:05:23 DEBUG[7768]: Local station id: Apr 19 21:05:23 DEBUG[7768]: Pages transferred: 1 Apr 19 21:05:23 DEBUG[7768]: Image resolution: 7700 x 7700 Apr 19 21:05:23 DEBUG[7768]: Transfer Rate: 9600 Apr 19 21:05:23 DEBUG[7768]: == Am I doing something wrong? Is my system doomed? ;) Julian J. M. ___ 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] TDM400P Revision question.
Hello, In FC3, i had to set wctdm options in /etc/modprobe.conf (it may be modules.conf in other distros): options wctdm boostringer=1 debug=1 Julian J. M. On 4/18/05, Ian Pattison <[EMAIL PROTECTED]> wrote: > 2. Low ringing voltage still (~44V AC). I have used the boostringer=1 option > when loading wcfxs, did I miss something at compile time? ___ 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] Call Files to Terminate a call to the dialplan not directly to a channel
You can use (at least in asterisk CVS), this: Channel: Local/[EMAIL PROTECTED] then in extensions.conf [from-internal] exten => 1234,1,Dial(whatever) exten => 1234,2,Dial(otherprov) Not testet though ;) Julian J. M. On 4/14/05, Mystery Glitch <[EMAIL PROTECTED]> wrote: > Can I use the .call files to place a call using the dialplan instead of the > channel directly? > > ---Channel: SIP/[EMAIL PROTECTED] > Context: testing > Extension: playsample > Priority: 1 > CallerID: Company <8882650946> > WaitTime: 15 ___ 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] Remote phone often appears to be disconnected
Just set qualify=yes in sip.conf On Apr 12, 2005 3:41 AM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: > Is there a possible settings for a remote SIP phone, so that a router > will not close the connection due to long time inactivity? ___ 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] Can you comment on this Qos script? How does one shape RTP?
You can shape incoming (TCP) traffic by dropping packets that exceed your download limit... But for this you rely on the other end to actually decrease their sending speed, i.e., if they knowingly flood you, there's nothing to do in your end.. The way to suggest to limit download speed by dropping ack packets may not work, or at least isn't as efective... In TCP there isn't an ACK for every incoming packet, they can be grouped, i.e., and ACK for tcp packet with seq=1000, actually acknowledges all packets until that sequence number, even if you dropped that ACK for seq=900... Correct me if i'm wrong ;) Julian J. M. On Apr 10, 2005 6:20 PM, cmisip <[EMAIL PROTECTED]> wrote: > 1. Qos is all about managing upload packets ( and download packets > indirectly by managing upload packets). ___ 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] Several INVITE messages sent by Asterisk
Try: canreinvite=no in your sip user definition. Julian J. M. On Apr 8, 2005 4:23 PM, Marlène Beray <[EMAIL PROTECTED]> wrote: > When I call from an IP Phone registered to the Asterisk server, the > connection is established and I can hear what the other person says but this > other person does not hear me. In fact, the Asterisk sends an Invite message > to the VoIP operator which replies; the connection is established. However, > the Asterisk sends another Invite to the firewall of the VoIP operator which > drops the message. As a consequence, the messages from the other person > reach the IP Phone but the messages sent by th IP Phone are dropped by the > firewall. ___ 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] Stopping Retransmission Found: 102 Error with Polycom IP300
I'm having this problem too, with a Swissvoice IP10... No nat between asterisk and the phone... I don't have any problems with the phone, outgoing and incoming calls work as expected... Could it be related to qualify=yes? Julian J. M. On Apr 6, 2005 1:39 PM, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: > Min Hwan Chang wrote: > > I'm having problems with a Polycom IP300 giving me a "Stopping > > Retransmission Found:102". It gives this error about every 30 > > seconds. ___ 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] Giving sip users multiple contexts
Create serveral contexts, e.g. from-sip-group1, from-sip-group2, etc... Then in that context, include the features you'd like for each group, and give each sip user the correct context. Julian J. M. On Wed, 30 Mar 2005 09:30:16 -0500, Matt <[EMAIL PROTECTED]> wrote: > How would I go about giving sip users multiple contexts? For instance > right now I have them all in: from-sip-internal > > Is there a way I can (for sip users) also include say my [dial-911] > [dial-local] and [dial-longdistance].. bearing in mind that I want to > have different sips allowed to do different things so I can't just do > includes for those in my from-sip-internal. ___ 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] problem with 1 dialing (recording says must dial 1 when I thought I did)
Maybe the first digit is dialed before the dialtone, try adding a 'w'
before ${EXTEN..., e.g.
exten => _91NXXNXX,2,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD1}})
Julian J. M.
On Mon, 28 Mar 2005 13:19:03 -0500, Kellner, Peter
<[EMAIL PROTECTED]> wrote:
> When I dial a long distance number (916503270309 for example) I get the
> message (I think from SBC) saying I must first dial a 1. Other times,
> it works, like when I dial this number (914082341389).
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Re: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
Have a look at http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision Julian J. M. On Mon, 28 Mar 2005 11:21:09 +, Robson Ribeiro <[EMAIL PROTECTED]> wrote: > After the call is finished if the user doesn't press # the line hangs forever. ___ 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] Using call.sample on Zap hardware - Answering problem
On Sun, 27 Mar 2005 12:29:55 -0500, Patrick Healy <[EMAIL PROTECTED]> wrote: > I've got a X100P connected to a POTS line and am using it to call out to > play a recorded message. I drop a copy of sample.call into > /var/spool/asterisk/outgoing and Asterisk picks up the line and initiates > the call. The problem is that the recorded message starts immediately and > doesn't wait for the called party to pick up the phone. When I try this > same process with a SIP extension, the process works like a champ, it just > fails on the Zap interface. This is normal, on analog Zap channels, asterisk doesn't know when the called party picked up the phone, unless your Telephone Company provides you with such information, usually via polarity switchs. You can enable debug on module wctdm (debug=yes), and watch /var/log/messages. Check if it detects polarity reversal when the called party picks up the phone, and when it hangs up. If it does, you could use answeronpolarityswitch=yes, and hanguponpolarityswitch=yes in zapata.conf. I've recently submited a patch to the bugtracker (http://bugs.digium.com/bug_view_page.php?bug_id=0003874), that fixes some problems with this approach (at least in Spain). Julian J. M. ___ 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] atxfer
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk <[EMAIL PROTECTED]> wrote: > I have installed asterisk 1.05 on debian sarge (binary package) > with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) > I am trying to get supervised/ attended tranfer working, blind transfer > by pressing the # key works fine > atxfer => * Attended transfers are only supported in CVS, not 1.0.X Julian J. M. ___ 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] Accecpt SIP calls from an IP
If you want to authenticate by IP, you need to add: insecure=very Julian J. M. On Tue, 15 Mar 2005 17:19:17 -, Kanishka Somaratne <[EMAIL PROTECTED]> wrote: > I want to enable SIP calls from an ip address, direct calling without > registering, the ip which sends the calls will not change. i have the > following in the sip.conf file > > [cisco4] > type=friend > host=192.168.0.5; This device registers with us > canreinvite=no ; Asterisk by default tries to redirect the > defaultip=192.168.0.5 > context=income ___ 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] sip.conf entry precedence
Try merging both and use type=friend Julian. On Sun, 13 Mar 2005 21:07:06 +0100, Pepe Aracil <[EMAIL PROTECTED]> wrote: > I only can get outgoing or incoming calls work well, but not both. > How can i solve this problem? ___ 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] Running asterisk as non-root: Zaptel Permission Probs
Why not chown to the user asterisk is running under? That way you don't give write access to everybody. AMP does that. Julian J. M. On Sun, 13 Mar 2005 13:31:12 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote: > As such, chan_zap is unable to work due to bad permissions. Is it safe to > simply change permissions on all /dev/zap/* stuff to rw-rw-rw ? Is there a > better/safer alternative? ___ 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] Need help on * anf HFC.
Hello,
I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context "from-pstn" if you want to receive calls.
group = 0
context=from-pstn
channel => 1-2
BTW, i'm from Spaintoo, and I'm really interested in knowing if your
setup works ;)
On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca <[EMAIL PROTECTED]> wrote:
> [channels]
> group = 1
> context=outbound-trunks
> channel => 1-2
> Mar 6 21:40:01 VERBOSE[21452]: -- Executing Dial("SIP/200-1cf6",
> "ZAP/g0/9639712471") in new stack
g0 means channel group 0, and you had group 1
Julian.
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Re: [Asterisk-Users] DyDNS + externip
Yes, you can, but asterisk needs to be reloaded (sip reload) when your ip changes. Julian J. M. On Thu, 3 Mar 2005 14:57:15 -0500, Giovanni Powell <[EMAIL PROTECTED]> wrote: > Can i use a domain name instead of an IP address for externip > (sip.conf) Because im using dynamic dns. Not sure what i'm trying to > achieve as yet but, i want to know if it is possible? ___ 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 URL and Callcenter Apps
It can be done with FOP (flash operator panel), which you can download from www.asternic.com. Also, FOP is included in AMP (Asterisk management portal) http://amp.coalescentsystems.ca/ Julian J. M. On Wed, 2 Mar 2005 15:39:32 -0600, Anton Krall <[EMAIL PROTECTED]> wrote: > How do those callcenter apps work with Asterisk where a call comes in and * > send a URL and some screen popup up based on callerid or something or > username or id and shows all the customers info? ___ 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] Sending Voicemail's to two email addresses
If you really need it, you can create an alias that send that mail to the addresses you want. Julian. On Thu, 03 Mar 2005 07:07:17 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote: > On Thu, 2005-03-03 at 06:32, Randy Johnson wrote: > > Is there a way to send a voicemail to two different email addresses when > > a caller leaves a message? > > Does "address1, address2" work or does it get confused about the ","? ___ 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] How to change fxo_mode
Hi, It's /etc/modprobe.conf in Fedora Core 3 ;) Julian J. M. On Wed, 2 Mar 2005 22:29:15 +0200, Soner Tari <[EMAIL PROTECTED]> wrote: > Thanks Julian, that's what I was looking for, and it worked of course. > (A note for google searchers: You mean /etc/modules.conf) > Soner ___ 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] How to change fxo_mode
Just add this to /etc/modprobe.conf: options wctdm opermode=TURKEY Julian J. M. On Wed, 2 Mar 2005 18:15:24 +0200, Soner Tari <[EMAIL PROTECTED]> wrote: > Sorry for littering the maillist, I've found it myself, I've changed the > wctdm.c file and make install'ed zaptel drivers, now it shows: ___ 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] More NAT questions
In you asterisk sip.conf: [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is If you don't externip, externip will never be used, because asterisk won't know WHEN to use it. Also, define canreinvite=no in your sip phones sections, as was suggested above. Julian J. M. On Wed, 2 Mar 2005 23:26:56 +1100, Rudolf Ladyzhenskii <[EMAIL PROTECTED]> wrote: > Hi, all > > Still trying to get NAT working. > > I have following setup: > > PHONE 1 -- * BOX > | > NAT/Firewall > | > | > NAT/Firewall >| >| > PHONE 2 > > Firewall next to phone 2 has all ports open. > Firewall next to Asterisk has open ports 5060 and 1:2. All of those > are forwarded to Asterisk box. > > Both phones succesfully register with Asterisk. (I had to add NAT=yes to > configuration of PHONE 2 in sip.conf to get this far). > Now, problems: > I can place a call from PHONE2 to PHONE1, but sound path is not established. > Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is > because port 5060 is not forwarded to the phone at NAT/Firewall, but more on > it later). > > Looking at SIP debug info, Asterisk tries to use local address of PHONE2 > instead of its public IP. As a result, no info can be sent to it. > > I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, > but this did not help. > > Now, we have tried to use one of the commercial VoIP service at PHONE2 > location. We had to use their phone and it worked just fine without any > alterations to NAT/Firewall device. I am pretty sure that they use SIP, so > they did resolve the problem somehow. Sorry, there is no technical info > available on this service. > > Did anyone succeeded in doing this setup? I know, IAX is a better way, but I > can not setup many Asterisk boxes. > > Basically, I am doing it for a friend. He is working for a small medical > company. They have number of offices that are not open every day and offices > are too small to put Asterisk box in each one. There will be 1-3 IP phones > in each office, except central one. Central one will need Asterisk, the rest > should be on their own. > > Any help is greatly appreciated. > > Thanks, > Rudolf > > ___ > 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
