[asterisk-users] SNOM M9 : expand range
Hello list, with the SNOM M9 DECT base station and handhelds, how can the range best be expanded ? Is there a DECT repeater that can be used ?? Is there a way to put some 'dumb' base station somewhere else on the network to expand the range ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to log into separate file
Thanks all, I guess i will use syslog for as my choice. Quyps On Thu, Apr 8, 2010 at 10:16 PM, David Backeberg dbackeb...@gmail.comwrote: On Wed, Apr 7, 2010 at 10:12 PM, Pham Quy qu...@vega.com.vn wrote: Hi all, I want to have a separate file to log what i need for my dialplan without all output from Asterisk. By this way, i can easily to trace problems caused by my dialplan. How can i do that? That's honestly a pretty vague question. Any number of problems could be caused by your 'dialplan'. syslog-ng It's nice. You can tune very specific statements to go to the arbitrary file of your choice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud Telephony
Today, Chris Matthieu, Founder CEO of GetVocal, entered the cloud-based communications market in February, with its launch of Teleku. Teleku is a new cloud-based telecom service that allows Web developers to build and host phone applications that answer inbound calls and initiate outbound calls, interact with Web applications, and send/receive SMS text messages! Chris, who you may have met at Astricon last year, is our guest later today. If you're interested in the cloud - and who isns't, even if you don'thave immediate plans - join us, first on IRC on Freenode.net (channel #vuc) or http://vuc.me/irc for a web-based client. The VUC takes place at Noon Eastern US Time, but for your time zone, look here : http://vuc.me/next General info on how to connect, etc: http://vuc.me SIP sip:200...@login./zipdx.com is best for g722 wideband-capable phones and accepts g711 as well You can also call sip:7463#2262...@proxy.ideasip.com to connect to the Talkshoe bridge. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR menu sound processing for AMR and GSM + live test available
On 04/09/10 05:08, Steve Edwards wrote: On Fri, 9 Apr 2010, Arkadi Shishlov wrote: It would be essential to get your comments (in email or by leaving a voice message) about sound quality if you could call the menu at sip:1...@riga.beta.lv (actually, any number at riga.beta.lv) I get: -- Executing Dial(SIP/501-0961b3a8, sip/1...@riga.beta.lv) in new stack -- Called 1...@riga.beta.lv -- Got SIP response 488 Not acceptable here back from 213.21.217.130 -- SIP/riga.beta.lv-09623508 is circuit-busy It was A-law only, on purpose. Just added U-law to the list. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
I really like the idea. I will try to ask. I don't know if they will be able to do that easily though. They ask a week or two for any changes to the hunt programming. Thanks, Bruce On Thu, Apr 8, 2010 at 3:29 PM, Edo edo.eku...@gmail.com wrote: Hello.. maybe you can just have the telco do an immediate forward of that number to the fifth number in the hunt group until it is fixed... On Thu, Apr 8, 2010 at 1:15 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 8 Apr 2010, John Novack wrote: A simple short on the pair will fix that, though that would require you to be on site, not always an option Would sacrificing a spare line cord (cut, strip, twist together) be an option for the on-site staff? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ekunwe EDO Network Services Tel: 601.497.3932 Fax: 601.500.6990 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] scratchy sound
Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can't determine the possible cause of this kind of distortion. Maybe an expert ear can give me a clue. It shouldn't be the Asterisk server's CPU usage. Could it be a network issue between the Asterisk system and the SIP client? (it happens with SIP hardphones as well as softphones so I guess it's improbable it's the client software/firmware) Both softphones and hardphones use GSM and usually work fine (this kind of issue is not too frequent). The LAN isn't dedicated to voice but has QoS prioritizing VoIP. Could the cause of the distortion be network-related? And only on my side? Should I consider other causes? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] run script after completed
Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] run script after completed
On 04/09/10 15:34, Necati Demir wrote: I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? You can write a script that watches outgoing_done with inotify for changes with minimal overhead. http://wiki.github.com/rvoicilas/inotify-tools/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] run script after completed
Do the call in a context and have the context run the script as a DeadAGI. [call_and_do] - exten = s,1,Dial. - exten = h,1,Deadagi(.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir Sent: Friday, April 09, 2010 7:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] run script after completed Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Ok... They normally do it here within a few hours.. This will not be a change to the hunt group just a forward immediately from one number to another. If the number was functional you could even have done it yourself using the forward code. On Fri, Apr 9, 2010 at 6:04 AM, bruce bruce bruceb...@gmail.com wrote: I really like the idea. I will try to ask. I don't know if they will be able to do that easily though. They ask a week or two for any changes to the hunt programming. Thanks, Bruce On Thu, Apr 8, 2010 at 3:29 PM, Edo edo.eku...@gmail.com wrote: Hello.. maybe you can just have the telco do an immediate forward of that number to the fifth number in the hunt group until it is fixed... On Thu, Apr 8, 2010 at 1:15 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 8 Apr 2010, John Novack wrote: A simple short on the pair will fix that, though that would require you to be on site, not always an option Would sacrificing a spare line cord (cut, strip, twist together) be an option for the on-site staff? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ekunwe EDO Network Services Tel: 601.497.3932 Fax: 601.500.6990 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ekunwe EDO Network Services Tel: 601.497.3932 Fax: 601.500.6990 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer
- dotnetdub dotnet...@gmail.com wrote: I would not think you'd need to worry about jitter on a normal 100mbit LAN unless there is heavy traffic or people are running their PC's through the phone (don't remember if the 501 has two ethernet ports...). Typically the quality issues are introduced on your WAN connectivity between the premise system and your hosted system aka 'The Big Internet'. Why do you mention running PC's through the phone like it will cause a problem? Do you normally like to put a network sensitive service like voice at the mercy of another device with the potential of chewing up the available bandwidth to the device? I personally do not like to share connectivity where possible. If you're asking why, then you have not experienced any issues with it yet. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound
Hi Vieri, The sound I hear does not seem caused by packet loss, jitter or latency, this problems usually produces a robotic or synthetic voice. It seems produced by some kind of bad contact (most probable). It is strange that you are seeing it using hard phones, I could bet on the headphones. Check also for electric interference, it is very common for line filters to break (broken capacitor usually) causing this kind of interference. Try also a different codec such as G.711 (improbable). Flavio E. Goncalves www.asteriskguide.com 2010/4/9 Vieri rentor...@yahoo.com Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can't determine the possible cause of this kind of distortion. Maybe an expert ear can give me a clue. It shouldn't be the Asterisk server's CPU usage. Could it be a network issue between the Asterisk system and the SIP client? (it happens with SIP hardphones as well as softphones so I guess it's improbable it's the client software/firmware) Both softphones and hardphones use GSM and usually work fine (this kind of issue is not too frequent). The LAN isn't dedicated to voice but has QoS prioritizing VoIP. Could the cause of the distortion be network-related? And only on my side? Should I consider other causes? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer
Do you seperate your voice and data networks? On 9 April 2010 14:56, Tim Nelson tnel...@rockbochs.com wrote: - dotnetdub dotnet...@gmail.com wrote: I would not think you'd need to worry about jitter on a normal 100mbit LAN unless there is heavy traffic or people are running their PC's through the phone (don't remember if the 501 has two ethernet ports...). Typically the quality issues are introduced on your WAN connectivity between the premise system and your hosted system aka 'The Big Internet'. Why do you mention running PC's through the phone like it will cause a problem? Do you normally like to put a network sensitive service like voice at the mercy of another device with the potential of chewing up the available bandwidth to the device? I personally do not like to share connectivity where possible. If you're asking why, then you have not experienced any issues with it yet. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer
- dotnetdub dotnet...@gmail.com wrote: Do you seperate your voice and data networks? Un-top-posting... Yes, I separate voice and data. Typically this is done using separate switches where possible, other times, using VLANs with appropriate QoS. Regardless, your phone and PC are sharing the same physical link to your switching infrastructure. If that works for you, great. It is not acceptable for me. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please sign Petition - Stop Child Labour
Are you sure writing to the right list??? Martin - Original Message - From: Sarfaraz Chougule To: sarfaraz.choug...@gmail.com Sent: Monday, April 05, 2010 4:54 PM Subject: [asterisk-users] Please sign Petition - Stop Child Labour Hello Friends, Kind request to you all - If you would want 6 crore children to have childhood please sign a petition on http://www.indyatweets.com (image on your top right) -- With Best Regards, *** Sarfaraz Chougule ***-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 69, Issue 16
On Thu, Apr 8, 2010 at 10:33 PM, Alan Zheng machinecat1...@gmail.com wrote: Hello All: I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO sample configure file for them. Is anybody know how to use them, or where is the documentation for them? If you read the code for those modules, you will learn there are NO sample configuration files because they are dialplan functions. See voipinfo for functions like: ReceiveFax() and ChanSpy() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
I am just guessing, but sometimes happened to me that the logic on dialplan
does not contain a hungup, so channels on spa3102 continues up even if users
have finished.
On CLI you should put core show channels, and see if there are channels to
sip/8028
On the [gw8028] context you send the call to [from-internal] extension 111,
so that extension has to end in a hangup action.
Just guessing.
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/4/9 Seann Clark nombran...@tsukinokage.net
Yes, the SPA-3201 is set as: (S0:8028) on dialplan 8, which is what I
have the device set to use. My bare bones working dialplan from Callweaver
works nearly perfectly with Asterisk, and takes all the calls and works just
as it did in Callweaver (making adjustments for the differences in dialplan
syntaxes as Callweaver still uses Asterisk 1.2 syntax). It is just after an
hour I can't get calls inbound to Asterisk. If I stop Asterisk, and start
Callweaver, it can sit for months and handle calls no problem, with a like
dialplan. SIP users and settings aren't changed between the systems either,
and my Cisco phones, and the other Linksys ATA I have plays well. I am a
little stumped on that. I will include a SIP dump when I get that back up in
test mode (Since it is my home telephone system and I need it for work,
which I am doing right now, I can't afford the downtime right this moment,
but tomorrow I should have time for this).
Thanks in advance,
Seann Clark
On 4/9/2010 12:08 AM, Jose Flores Galicia wrote:
Hi.
On the Spa 3102 is set as Dialplan s0:8028 on PSTN line tab, since other
way the incoming call will try to be routed to a non set extension on
[gw8028] context
Best Regards
Jose Flores Galicia
floj...@gmail.com mailto:floj...@gmail.com
BriefCode Code Based Training
2010/4/8 Seann Clark nombran...@tsukinokage.net mailto:
nombran...@tsukinokage.net
All,
I am looking at a little support on this, as I haven't found it
on google yet. I have had this work on Callweaver, but am moving
to Asterisk for a variety of reasons. My dial plans, and
everything else transferred perfectly, though I am not sure they
are 'correct' for Asterisk 1.6.1, with simple things like SIP
users outlined in the sip.conf file, not in the users file, and my
dialplan syntaxes don't appear to be liked by the asterisk-gui
program (not a big deal, was just something shiny to look at for
me, to try to figure out a way to get this going).
What my problem is with Asterisk is my SPA-3201 is my primary
voice gateway, as I do not own any Digium hardware, and currently
do not have a SIP provider outside of my PBX at home. When I
restart Asterisk, everything works perfectly. I let Asterisk sit
for an hour or so, and it stops allowing calls to be routed into
the assigned extension. I do see stuff from the communications, at
the time the call lands on the Asterisk server:
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
The logic is that the SPA is registered as an extension on my
system, and incoming calls are routed into the system VIA that
extension. The dialplan that the SPA connects to is:
[gw8028]
exten = 8028,1,Answer
exten = 8028,n,Set(CallerNum=${CALLERID(num)})
exten = 8028,n,Set(CallerName=${CALLERID(name)})
exten = 8028,n,Set(CDR(accountcode)=8203)
exten = 8028,n,Set(CDR(UserField)=POTS)
exten = 8028,n,Goto(from-internal,111,1)
exten = 8028,n,Hangup
the 'from-internal' is my current call filtering/processing subsystem.
The outbound side of this works just fine though, as well as my
ATA's and Cisco 7960's are able to make and receive calls when
this is happening. I can include any additional details if
requested, as I don't know exactly what would be helpful to others
with this. The SPA itself hasn't been changed in seven months, and
is stable with Callweaver.
Thanks in Advance,
Seann Clark
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Re: [asterisk-users] scratchy sound
Am 09.04.2010 13:10, schrieb Vieri: Please listen to the following sound file: I've experienced similar (well, vaguely similar) distortion on a horstbox pro when echo cancellation is switched on for the zap channels (ISDN). Turning it off resulted in no distortion at all, but then i experienced very strong echo at times. Switching on echotraining lowered the distortion so that's currently the trade-off i have to live with. HTH a little. -- -- o -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound
Hi, sounds for me like when i use an headset and the microfone handle scratches on my beard while i talk ;) maybe you have a network cable whitout screening. I had bad problems on different phones which sounds like that you have cause of electric or magnetic inteferences but when i changed the network cable everything was fine again. do you have made an echo test lie exten = 999,1,Echo() to hear yourself and try to find this issue. best regards steve smith Vieri schrieb: Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can't determine the possible cause of this kind of distortion. Maybe an expert ear can give me a clue. It shouldn't be the Asterisk server's CPU usage. Could it be a network issue between the Asterisk system and the SIP client? (it happens with SIP hardphones as well as softphones so I guess it's improbable it's the client software/firmware) Both softphones and hardphones use GSM and usually work fine (this kind of issue is not too frequent). The LAN isn't dedicated to voice but has QoS prioritizing VoIP. Could the cause of the distortion be network-related? And only on my side? Should I consider other causes? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid over IAX Trunks
Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the extension and password are 999 2. I then created IAX trunks for each box using 999 as username and password, hostname/IP was set to be other box's IP 3. when done, from the system status panel, I saw the trunks successfully registered to the other box 4. then I added Outgoing Call Rules to each box: for box1, _2XX -- to_box2_trunk for box2, _1XX -- to_box1_trunk This setup works ok, the only problem is caller id, i.e. when extension(200) from box2 calls to extension(100) from box1, the call can be made but the caller id displayed on 100 is 999 not 200. I have been on this problem for some time already, could anyone here give me a bit help please? -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Timezones
Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2 timezone. I have looked around and I do not have found any information about how to set the log/system timezone. The only place I remember having a reference to timezones is the voicemail config file; but I do not get the link to 'server time'. Any idea? Tia, Aldo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer
On 9 April 2010 16:46, Tim Nelson tnel...@rockbochs.com wrote: - dotnetdub dotnet...@gmail.com wrote: Do you seperate your voice and data networks? Un-top-posting... Yes, I separate voice and data. Typically this is done using separate switches where possible, other times, using VLANs with appropriate QoS. Regardless, your phone and PC are sharing the same physical link to your switching infrastructure. If that works for you, great. It is not acceptable for me. --Tim I was asking the question as I was talking to a guy about this very same issue recently. He was adamant that you should never ever plug the PC into the ethernet port on a VOIP handset yet he had everything going back to a really bad dlink switch. There is a switch port in most good VOIP phones. I have seen large telcos in the UK using the switchport in their Cisco handsets. We have many installs doing the same thing with both Asterisk and Call Manager. Never has it caused any difficulty. Of course ymmv and all that. Cheers, S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res fax help
I have res_fax setup and working for the most part. However, I'm seeing some fax machines drop the connection on me - Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel 'DAHDI/1-1' did not return a frame; probably hung up. -- Channel 0/1, span 1 got hangup, cause 102 -- Channel 'DAHDI/1-1' FAX session '20' is complete, result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x98', transfer rate: '14400', remoteSID: 'numberredacted' == Spawn extension (macro-fax_rcv, s, 3) exited non-zero on 'DAHDI/1-1' in macro 'fax_rcv' It appears to be dropping out of my macro fax_rcv at that point and not executing the next step in the dialplan, which is a System call to a script that converts the tif to a pdf and emails it to the extension owner. My question is how do I ensure that my script is called when the far end hangs up before the call progresses that far in the dialplan? My first thought is to add something like this- exten = h, 1, System(callmyscript.sh,arg1,arg2,arg3,arg4,argimapirate) to the macro, but I'm not sure if that would do it or not. Anyone have any thoughts? Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with Fax over TDM410P
Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case! take a look in our zapata: [channels] language=es ;context=default rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes immediate=no ;busycount=4 ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ; TDM410P context = mde-g1 immediate=no signalling=fxs_ks group=0 channel = 1 context = mde-g1 immediate=yes Signalling=fxs_ks group=0 channel = 2 context = mde-g1 immediate=yes signalling=fxs_ks group=0 channel = 3 context=inside faxdetect=incoming immediate=no signalling=fxo_ks group=1 channel = 4 What should we do in order to make it work ok? we really need to put this working, i've heard that asterisk does not work very well with fax, but at least it should try to dend it, not to get frozen :S Thanks in advance for all your help! Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] softphone help
I am having serious problems connecting my client software to asterisk, i tried x-lite would not connect, and i tried with twinkle too, it wouldnt, i cannot get to call myself, i am not on a network, just trying all this out locally, can i not get to connect without been on a network? _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant send faxes to the pstn (through E1 PRI) and viceversa... What should we do to make this work properly? what parameters in zapata? mediant 1000? Thanks in advance for all your help! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant send faxes to the pstn (through E1 PRI) and viceversa... What should we do to make this work properly? what parameters in zapata? mediant 1000? Thanks in advance for all your help! I've had fairly good success with faxing using Asterisk + Hylafax. I haven't tried any of the built-in Asterisk faxing programs yet because I designed this setup before the newest revisions, when Asterisk + built-in faxing was not working well. What I do is run Hylafax on the same machine as Asterisk, and then run IAXModem to do the communication between the 2. There's a lot of documentation online about how to set this up. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone help
On Fri, 9 Apr 2010, ayodele abejide wrote: I am having serious problems connecting my client software to asterisk, i tried x-lite would not connect, and i tried with twinkle too, it wouldnt, i cannot get to call myself, i am not on a network, just trying all this out locally, can i not get to connect without been on a network? 0) A more specific subject will get more specific answers. 1) The more detail you provide, the more detailed answer you may receive. 2) Focus on a single issue. You can run a softphone on the same host as Asterisk if you configure the port usage correctly. It's easier if you use a separate computer (even a Windows computer) so you have less to figure out before you complete your first call. A complete, completely insecure sip.conf file can be as simple as: [general] disallow = all allow = all allowguest = yes allguest= yes context = block-ani host= dynamic ; (end of sip.conf) Note that you will have to change the context to match where you want the call to start in your dialplan (extensions.conf). You can view the SIP dialog between your softphone and Asterisk if you enable SIP debugging. For version 1.2, the CLI command is sip debug but you may be running a different version. If you still need help: ) start a new thread with a more meaningful subject like: First time user -- x-lite not connecting. ) Specify which version of Asterisk you are using. ) Cut and paste the contents of sip.conf. ) Cut and paste the output from the CLI command show dialplan the-name-of-the-context-from-sip.conf ) Enable SIP debugging using the appropriate CLI command for your version of Asterisk. ) Cut and paste the CLI output from a failed call attempt. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
