Re: [asterisk-users] High load on asterisk servers
Hi, I never see high load with only a few calls - but I'm not monitoring the servers during night where we have few calls. But we can get high load at ~30 calls also. /Henrik On 12/20/13 09:54 AM, Thorsten Göllner wrote: What about the load, when only 1 or 2 calls are on this machine? Am 20.12.2013 09:01, schrieb Henrik Andresen: Hi Stefan, I use own dns-servers on local subnet so I don't think it's the problem :( Also I have hosts in local hosts-files. /Henrik On 19/12/13 14:47, Stefan Schmidt wrote: Maybe this happens if you have a short delay to your dns servers. This could increase the load very fast and after some seconds it might be over again. I have installed a dns recurser with own caching on all of my asterisk servers and now everything runs much more smoothly. best regards stefan Am 19.12.2013 11:56, schrieb Henrik Andresen: All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] High load on asterisk servers
No :( Nothing to see in asterisk logs/console /Henrik On 12/20/13 09:55 AM, Thorsten Göllner wrote: Do you see no hint in the atserisk console (log)? Am 20.12.2013 09:01, schrieb Henrik Andresen: Hi Stefan, I use own dns-servers on local subnet so I don't think it's the problem :( Also I have hosts in local hosts-files. /Henrik On 19/12/13 14:47, Stefan Schmidt wrote: Maybe this happens if you have a short delay to your dns servers. This could increase the load very fast and after some seconds it might be over again. I have installed a dns recurser with own caching on all of my asterisk servers and now everything runs much more smoothly. best regards stefan Am 19.12.2013 11:56, schrieb Henrik Andresen: All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] High load on asterisk servers
Hi Stefan, I use own dns-servers on local subnet so I don't think it's the problem :( Also I have hosts in local hosts-files. /Henrik On 19/12/13 14:47, Stefan Schmidt wrote: Maybe this happens if you have a short delay to your dns servers. This could increase the load very fast and after some seconds it might be over again. I have installed a dns recurser with own caching on all of my asterisk servers and now everything runs much more smoothly. best regards stefan Am 19.12.2013 11:56, schrieb Henrik Andresen: All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] High load on asterisk servers
I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik -- _ -- 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] High load on asterisk servers
All calls are sip--sip On 19/12/13 11:32, Thorsten Göllner wrote: Am 19.12.2013 10:37, schrieb Henrik Andresen: I have a problem with asterisk. I got ~15 asterisk servers on new hardware (1 or 2 xeon 3ghz) sometimes I got high load between 1 and 10. No disk activity, no ram or swap problem. But asterisk main process is using up to 300-500% cpu. This happens both with 30 channels in use and 100+ channels in use. I'm not doing transcoding or anything. any clue ? One server with 300 channels load on 5 One server with 600 channels load on 0.02 After 5 minutes it might be ok... some times its ok after 1 hour. I do no recording, no transcoding just g711a Two servers does not have sip-registrations as they are gateways to our sip-propvider. The other servers got around 1000-1200 sip registrations. Running asterisk 11.5.x, 11.6.0 and now trying 11.7.0 All servers HP with centos 6.5 (has been 6.3 and 6.4 as well) Any clue ? /Henrik What calls cause these problems? SIP or E1/T1-Calls? -- _ -- 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] Problem with Caller ID when receiving hidden number in via DAHDI and redirecting out via SIP
Hi,
We have a system with both ISDN trunks and SIP. We receive incoming calls on
both but always dial out via SIP.
When dialing out the caller id is set like this:
exten = _X.,1,Set(CALLERID(num)=${CC_ORIGNUM})
exten = _X.,n,Set(CALLERID(name)=${CC_ORIGNAME})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, em)
This always works fine on SIP and on ISDN as well when the number is not hidden.
But for some reason the setting of the caller id does not work when receiving
calls from hidden numbers.
The from address in the outgoing SIP looks like this:
From: Anonymous sip:anonymous@anonymous.invalid
Does anyone know why this is happening, is there a way to go around it?
Regards,
Henrik
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Re: [asterisk-users] executing the h extension at the real hangup of the call
Ok, yes I find that strange as well. I will perform some tests on another
server.
/Henrik
Från: Gareth Blades
mailinglist+aster...@dns99.co.ukmailto:mailinglist+aster...@dns99.co.uk
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Datum: fredag 13 september 2013 13:53
Till: Asterisk Users Mailing List - Non-Commercial Discussion
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Ämne: Re: [asterisk-users] executing the h extension at the real hangup of the
call
On 13/09/13 12:31, Henrik Westerberg wrote:
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always
over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
I have no idea why you are seeing the h extension being run before the call
ends. Its not something I have ever seen happen.
Whether or not Asterisk stays in the RTP media path makes no difference as it
will always stay in the SIP signalling path and its that which controls the
call establishment and termination.
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[asterisk-users] executing the h extension at the real hangup of the call
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always
over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
Regards,
Henrik
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Re: [asterisk-users] Recording with MixMonitor and AGI
Hi, Ok, thanks. /Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 14 mars 2013 10:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, the music heard by MoH is configurable... so if you want silence... But hold could e.g. also be done by transferring a caller into a dynamic meetme room... yves Am 14.03.2013 08:43, schrieb Henrik Westerberg: Hi, The idea was to record an ongoing call by three party bridging on the mobile phone. Well my problem was to halt execution of the Dialplan so the server would not hang up the call. And I don´t want the server to say anything during the call. Now I solved this case as well by using Answer and then Record in the dialplan . So I´m not recording with MixMonitor. But just out of curiosity. How did you mean using hold (in answer/hold). Is that MusicOnHold? For me I can´t use that since I don´t want to make any noise. Is there another way? exten = 111,1,Answer() exten = 111,n,? I have tried using Wait with a long duration but have not succeeded to make it work as I want. I am using asterisk-java and originate calls to local channels. Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@ain.se Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record
Re: [asterisk-users] Recording with MixMonitor and AGI
Hi, The idea was to record an ongoing call by three party bridging on the mobile phone. Well my problem was to halt execution of the Dialplan so the server would not hang up the call. And I don´t want the server to say anything during the call. Now I solved this case as well by using Answer and then Record in the dialplan . So I´m not recording with MixMonitor. But just out of curiosity. How did you mean using hold (in answer/hold). Is that MusicOnHold? For me I can´t use that since I don´t want to make any noise. Is there another way? exten = 111,1,Answer() exten = 111,n,? I have tried using Wait with a long duration but have not succeeded to make it work as I want. I am using asterisk-java and originate calls to local channels. Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@ain.se Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although there is no absolute need for using an agi... you can all write down in your dialplan...) is there a special protocol requirement for saving/transferring the recorded voicefile (e.g. ftps)? One obstacle is, that the recorded file is not fully
Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
Thanks for your answer!
1.
so you want to establish a call (triggered by ami) between two partys, record
the conversation
and save the file to a(nother) server (afterwards), right?
Yes this is correct, and I prefer to do the transferring of the file to another
server with my existing AGI.
My AGIs are written in java. Today I the upload is done over http.
Today I schedule the upload in the AGI script a couple of seconds after the
channel is hang up. But the two
lines might not be hung up at the same time.
Your suggestion of always fixing the file is wise, it now seems to work fine
after having been processed with sox.
So now I think that this case 1 is ok for me :-)
2.
and another task is to establish (also ami triggered) a call to a mobile and
play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) server
(afterwards), right?
The idea is to perform a probe call with the only task of recording what the
other party says.
It will be merged by hand on a mobile phone to an ongoing call with another
party.
This could be done by calling out and letting AGI execute a RECORD FILE but if
there is a way to just
dial out and then let the server side of the call Keep the channel up but do
nothing forever until the call is hang up
Then I could easily use the MixMonitor and write the whole conversation in the
dialplan with uploading similar to the first case.
Any suggestions?
Regards,
Henrik
Från: Yves A. yves...@gmx.demailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
hi,
hard to understand, what your objective is... at least for me ;-)
so you want to establish a call (triggered by ami) between two partys, record
the conversation
and save the file to a(nother) server (afterwards), right?
and another task is to establish (also ami triggered) a call to a mobile and
play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) server
(afterwards), right?
let me know, if i understood you right, the solution is not so hard to
implement.
In what language do you preferrably write your AGIs? (although there is no
absolute need for using an
agi... you can all write down in your dialplan...)
is there a special protocol requirement for saving/transferring the recorded
voicefile (e.g. ftps)?
One obstacle is, that the recorded file is not fully written _immediately_
after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be
interrupted, if the call is hungup...
but as you did not show your agi... these are just hints..
regards,
yves
Am 07.03.2013 16:21, schrieb Henrik Westerberg:
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.zmailto:EXTEN}@x.y.z)
[outgoing-originate-rec]
exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten =
_X,n,Dial(SIP/${EXTEN}@x.y.z,60,Mmailto:EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I can cope
with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to upload
the file. The file will not have a duration. It works when I schedule the
uploading a while after from my agi application but I
Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
Ok but when I use the macro the recording doesn´t start until the call is
answered which is a plus. It´s easy to trim away silence of course though.
But according to the documentation it seems like DeadAgi is obsolete in
Asterisk 1.6 and later, that AGI should be used instead.
Regards,
Henrik
Den 2013-03-08 05:30 skrev Bharat Lalcheta bharatlalch...@gmail.com:
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
I am developing a call recording application on Asterisk 11.2 and have
this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten =
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is
${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to
08 I
then originate a call via AMI to Local/0@outgoing-originate with
context set to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed,
returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I
can
cope with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to
upload the file. The file will not have a duration. It works when I
schedule
the uploading a while after from my agi application but I would rather
not
rely on a timeout.
When I tried to run StopMixMonitor before the Agi call in the h
extension,
the first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited
non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to
specify a
command to execute when MixMonitor stops but I would rather handle the
file
uploading in my agi application.
I also have another case: I want to dial out a call and record it. It
will
be a oneway-call from the server to a mobile. Do I need to get
AGI-control
of it and record with an AGI command or how can I hack it directly in
the
dial plan using MixMonitor?
Best Regards,
Henrik
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[asterisk-users] Recording with MixMonitor and AGI
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I can cope
with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to upload
the file. The file will not have a duration. It works when I schedule the
uploading a while after from my agi application but I would rather not rely on
a timeout.
When I tried to run StopMixMonitor before the Agi call in the h extension, the
first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to specify a
command to execute when MixMonitor stops but I would rather handle the file
uploading in my agi application.
I also have another case: I want to dial out a call and record it. It will be a
oneway-call from the server to a mobile. Do I need to get AGI-control of it
and record with an AGI command or how can I hack it directly in the dial plan
using MixMonitor?
Best Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8
/H
Den 2013-01-02 22:25 skrev Danny Nicholas da...@debsinc.com:
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination
and we announce the call before the transfer is complete. The carrier
requires a diversion header if the ANI is not one of our DIDs. Does
someone have experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a
SIP Header command before the Dial to resolve the diversion header
issue.
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[asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,…)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=……..)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have AGI
control of it. Is there a way to do this or do I have to use G() and connect
the both ends to AGI separately and then bridging them before recording the
call?
Thanks for help.
Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
Thanks Danny I will try this.
/Henrik
Message: 12
Date: Wed, 2 Jan 2013 08:17:59 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Dialing out and recording
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 001501cde8f3$f7d2b290$e77817b0$@debsinc.com
Content-Type: text/plain; charset=us-ascii
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and
have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and
we
announce the call before the transfer is complete. The carrier requires
a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
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Re: [asterisk-users] Problem setting for incoming termination
for start you could disable guest access in sip.conf, I guess you do not
need it
On 2011.08.11 14:29, Jim Boykin wrote:
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
[dallas]
type=user
username=dallas
secret=somepassword
host=dynamic
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
accountcode=411
context=dallas
This is the configuration at client end.
[outgoing]
type=peer
username=dallas
secret=somepassword
host=ipaddress
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
We do not require the client to register, neither we want them to use
fromuser field. I think we are doing some silly mistake since this
should be a simple configuration used by many. Please help
Thanks
Jim
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Re: [asterisk-users] Firewall Issue
Also you can set allowguest=no in sip.conf, if you didn't do it already On 2011.08.08 13:24, RSCL Mumbai wrote: Hi, (1) Since a few days, I am seeing unexpected (unwanted) calls reaching my asterisk server. Please see attached log files. (2) I believe the source IP of these calls is the IP mentioned under the CHANNELS column. (3) But as per my firewall, these calls should not have reached Asterisk. The should have been dropped by the Firewall. Please suggest if my thinking is in the correct direction, and what should be my next step. Best regards, Sans -- _ -- 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] applicationmap and ChannelRedirect
Hi, Does the lack of answers prove that the problem described in bug 17117 isn't a problem in reality and everything is caused by my setup? /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26hemsida: www.whatever.nu -- _ -- 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] applicationmap and ChannelRedirect
* Zeeshan Zakaria zisha...@gmail.com [100620 16:14]: But I can tell you that I have implemented a couple of times somewhat similar dialplan functionality, recent of which was about 6 months ago. I used the n-way calling example as a reference and first made a working n-way feature, which itself was tricky, and then modified it for fit my requirement, which again, took quite a bit of debugging to make it work. So yes, I think it is somewhere in your dial plan that you should look. Have you tried to implement the n-way calling feature? That helped me a lot to understand how to successfully bridge in-progress calls to other channels, and make dynamic feature codes to work with it. Thanks for your answer. I haven't tried the n-way example, just looked at the dialplan and I think I have understood how it works. Will try the n-way example to see if I have missed something. Still the bug 17117 rings in the back of my head because it matches my problem... /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26hemsida: www.whatever.nu -- _ -- 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] applicationmap and ChannelRedirect
-- SIP/201-0006 is ringing -- SIP/201-0006 answered SCCP/203-0007 -- SCCP: Outgoing call has been answered SCCP/203-0007 on 2...@sep001121d89b97-0007 == Using SCCP RTP TOS bits 184 == Using SCCP RTP CoS mark 5 -- Feature Found: internal-move exten: internal-move -- Executing [...@macro-internal-move:1] ChannelRedirect(SIP/201-0006, SCCP/203-0007,internal-move-conference,blafs,1) in new stack -- Executing [...@macro-internal-move:2] NoOp(SIP/201-0006, Sent caller to dynamic conference) in new stack -- Executing [...@macro-internal-move:3] NoOp(SIP/201-0006, - hung up) in new stack == Spawn extension (internal-move-conference, blafs, 1) exited non-zero on 'SCCP/203-0007' -- Executing [bl...@internal-move-conference:1] Answer(SCCP/203-0007, ) in new stack -- Executing [bl...@internal-move-conference:2] Set(SCCP/203-0007, TIMEOUT(absolute)=10) in new stack Channel will hangup at 2010-06-17 12:09:03.167 CEST. -- Executing [bl...@internal-move-conference:3] MeetMe(SCCP/203-0007, 424242,d1MFqAx) in new stack -- Created MeetMe conference 1023 for conference '424242' -- Started music on hold, class 'default', on SCCP/203-0007 -- Stopped music on hold on SCCP/203-0007 -- Hungup 'DAHDI/pseudo-913977795' == Spawn extension (internal-move-conference, blafs, 3) exited non-zero on 'SCCP/203-0007' -- SCCP: Asterisk request to hangup channel SCCP/203-0007 -- SCCP: Request to schedule delete for channel '7' in 10 seconds -- SEP001121d89b97: Accessory 'Speaker' is 'OnHook' (0) -- SEP001121d89b97: Statistics from 201 callid: 7 Packets sent: 93 rcvd: 88 lost: 1 jitter: 0 latency: 0 /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26hemsida: www.whatever.nu -- _ -- 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] async agi question
Thanks, I was not familiar with this application. /Henrik Kevin P. Fleming skrev: Henrik Westerberg wrote: Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. The ExternalIVR interface, while not quite as feature-filled as AGI, does in fact work in a true non-blocking fashion, and supports exactly what you are looking for. In fact, needing to be able to stop playback of prompts asynchronously was the primary reason it was developed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] async agi question
Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command asyncagi break does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik ___ -- 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] async agi question
Hi Moy, Thank you for your quick answer. Also thanks for implementing the great async agi functionality! Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. Regards, Henrik Moises Silva skrev: Hello Henrik, I have not used Asterisk from a user perspective lately, but, when I added the async agi functionality, I used to control this using a "manager redirect" action to the same priority where the channel calls async agi, that will work like a break that re-enters the async agi loop . This, of course, requires you to save the state of the channel somehow in your program to "remember" that the next time that channel calls async agi the sound was already played and such. http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Let me know if that does not work for you and we can probably write something in res_agi.c Moy On Fri, Dec 5, 2008 at 3:01 AM, Henrik Westerberg [EMAIL PROTECTED] wrote: Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command "asyncagi break" does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik ___ -- 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 -- Med vnliga hlsningar / Best Regards Henrik Westerberg Software Developer Aurora Innovation AB Vallongatan 1, 752 28 Uppsala, Sweden direct: +46 18 19 44 58 mobile: +46 703 28 98 40 email: [EMAIL PROTECTED] www.aurorainnovation.se ___ -- 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] SIP-Realtime and sip reload
Am Donnerstag, den 06.12.2007, 21:06 +0100 schrieb Torbjörn Abrahamsson: Our current approach is to use the #exec directive, and call a script which creates static friends by reading information from the DB. We still use the remote ITSP peers with realtime, as they do not need the OPTIONS. This way when we call a reload the users registration is still there, and we have the flexibility of using a DB as the user database. Could you explain that a litte bit to me? I just tried to find something about #exec, but not very successfully. Is there any documentation? Do you reload asterisk and generate the sip.conf by reading the users from a database with a script? And omit the usage of realtime for these users? Could you perhaps post/send your configuration/script? thanks Henrik ___ --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] SIP-Realtime and sip reload
Am Mittwoch, den 05.12.2007, 17:14 -0600 schrieb JR Richardson: I use SIP-Realtime to store my SIP-users and I keep the informations about the SIP-Providers my Asterisk registers to in sip.conf. I'm running into the following problem. If I set rtcachefriends=yes because I want to use MWI and run a sip reload because I changed something in sip.conf, Asterisk forgets about all registrations of the users which are all unavailable after that. How can I use rtcachefriends=yes to allow MWI (isn't it needed for NAT-keepalive as well?) and don't break everything with a sip reload? The short answer is, this is how it works, don't reload sip.conf or loose your cache. You can set your phone registration time lower that 3600 so phones re-register quicker. Yes, that was my workaround which might be a solution unless you do a reload on the busiest time of the day ;-) Thanks for the clarification. Henrik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP-Realtime and sip reload
Hi, I use SIP-Realtime to store my SIP-users and I keep the informations about the SIP-Providers my Asterisk registers to in sip.conf. I'm running into the following problem. If I set rtcachefriends=yes because I want to use MWI and run a sip reload because I changed something in sip.conf, Asterisk forgets about all registrations of the users which are all unavailable after that. How can I use rtcachefriends=yes to allow MWI (isn't it needed for NAT-keepalive as well?) and don't break everything with a sip reload? thanks for your help Henrik ___ --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] BeroNet HFC-4S card is now detected as only 2 ports
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Output of lspci -vvv
-- cut
02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Unknown device b560
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 22
Region 0: I/O ports at ddb8 [size=8]
Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
-- cut
Just a comment: the CHANGES file has the item fixed detection of
miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c
in -d and in -e .
Problem solved...
If anyone else is interested, here is what I changed to make it work
with a BeroNet HFC-4S rev 01 card:
Patch file:
cut
297a298,299
} else if (qoztmp-type == 0xb560) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
1584c1586
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
---
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
1638a1641
case 0xb560:
cut -
I don't know how to make it into a correct patchfile, so if someone else
knows that, it could be done and maybe placed where everybody could get it.
---
Best regards,
Henrik Woffinden
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Re: [asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports
Tzafrir Cohen wrote:
On Sat, Apr 07, 2007 at 12:17:03PM +0200, Henrik Woffinden wrote:
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it only sees 2 ports.
Output of lspci -vvv
-- cut
02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Unknown device b560
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Interrupt: pin A routed to IRQ 22
Region 0: I/O ports at ddb8 [size=8]
Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA
PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
-- cut
Just a comment: the CHANGES file has the item fixed detection of
miniPCI cards (qozap). Take a look at the diff between qozap/qozap.c
in -d and in -e .
Problem solved...
If anyone else is interested, here is what I changed to make it work
with a BeroNet HFC-4S rev 01 card:
Patch file:
cut
297a298,299
} else if (qoztmp-type == 0xb560) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
1584c1586
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
---
if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
1638a1641
case 0xb560:
cut -
I don't know how to make it into a correct patchfile, so if someone else
knows that, it could be done and maybe placed where everybody could get it.
Simply use 'diff -u' instead of 'diff' .
Any idea if this can break anything?
Thanks for the tip. Here's a proper patch-file:
[EMAIL PROTECTED] qozap]# diff -urN qozap.c.orig qozap.c
--- qozap.c.orig2007-04-07 11:49:36.0 +0200
+++ qozap.c 2007-04-07 12:00:52.0 +0200
@@ -295,6 +295,8 @@
} else {
if (qoztmp-type == 0x08b4) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x0);
+ } else if (qoztmp-type == 0xb560) {
+ qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
} else if (qoztmp-type == 0xb550) {
qoz_outb(qoztmp,qoz_R_BRG_PCM_CFG,0x23);
} else if (qoztmp-type == 0xb556) {
@@ -1581,7 +1583,7 @@
if (pcidid == PCI_DEVICE_ID_CCD_M) {
qoztmp-stports = 8;
} else {
- if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558)) {
+ if ((tmp-subsystem_device = 0xb555) ||
(tmp-subsystem_device == 0xb558) || (tmp-subsystem_device == 0xb560)) {
qoztmp-stports = 4;
} else {
qoztmp-stports = 2;
@@ -1636,6 +1638,7 @@
if (pcidid == PCI_DEVICE_ID_CCD_M4) {
switch (tmp-subsystem_device) {
case 0x08b4:
+ case 0xb560:
if (ports == -1) ports = 0; /* assume TE mode if
no ports param */
printk(KERN_INFO
qozap: CologneChip HFC-4S evaluation board
configured at io port %#x IRQ %d HZ %d\n,
I don't think it will break anything, as I haven't changed any logic,
just added the device 0xb560 to go to the proper options. But I can't
give u a 100% guarantee as I have no experience in programming these cards.
Shall I upload the patch to somewhere?
Happy Easter.
Best regards,
Henrik Woffinden
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[asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports
Hello list, After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently detected as 2 ports instead of 4. I still load the driver as modprobe qozap ports=12 as I've always done. But now it only sees 2 ports. Output of lspci -vvv -- cut 02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b560 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at ddb8 [size=8] Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- -- cut Anyone has any ideas? -- Best regards, Henrik Woffinden ___ --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] Problems with mISDN TE line
Hi list, I've installed Asterisk 1.4.0 with newest mISDN 1.0.4 + mISDNuser 1.0.3 on Fedora Core 6. I get many compilation error on mISDN. It wants to include linux/config.h That I fixed by removing the #include line at every occurance. (Don't know if that was a wise move, but it then compiled). mISDNuser and asterisk compiled fine, and asterisk can find and use the ISDN BRI port in nt_pmtp mode, but when I want to call out via the te_pmtp port, then the following line is written in the log: For incoming call: Sat Jan 13 16:33:04 2007: P[ 1] Extension can never match, so disconnecting For outgoing call: Sat Jan 13 16:41:18 2007: P[ 1] -- we have already send Release_complete and the line is BUSY. The phones on the NT line works fine, and SIP - mISDN (NT) calls has no problems. All comment lines has been removed to keep mail size reasonable. /etc/misdn-init.conf: card=1,0x4,dtmf te_ptmp=1,2 nt_ptmp=3,4 option=1,master_clock poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=1 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log bridging=yes stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=default language=en musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix= internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=yes reject_cause=16 need_more_infos=no nttimeout=no method=standard localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no always_immediate=no nodialtone=no immediate=no hold_allowed=yes callgroup=1 pickupgroup=1 presentation=-1 screen=-1 echocancel=yes echocancelwhenbridged=yes echotraining=yes jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=yes max_incoming=-1 max_outgoing=-1 [outside] ports=1 context=outside msns=* [inside] ports=3 context=inside msns=* overlapdial=yes -- Med venlig hilsen / Best regards, Henrik Woffinden ___ --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] Digium Wildcard B410P
Hi list, Is Digium Wildcard B410P compatible with Asterisk 1.2 / 1.4 (zaptel of course) directly out of the box, or do I need things like bristuff? http://www.digium.com/en/products/hardware/b410p.php Best regards, Henrik Woffinden ___ --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] Upgrading from 1.2.12.1 to 1.2.13
Hi, After upgrading from: Zaptel 1.2.9.1 Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s to Zaptel 1.2.10 Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v I get the following error when connecting my Xlite Softphone: --- cut --- Nov 4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of 0x886df58 (len 486) to 192.168.9.9:31308 returned -1: Operation not permitted --- cut --- It seems to be Xlite wanting to see who of my contacts is on-line. There's no problem phoning, but all my contacts are offline according to Xlite. sip show peers on the CLI tells me different. There are hint lines for everybody. And it worked perfectly in 1.2.12.1 Does anyone know what the error could be? -- Med venlig hilsen / Best regards, Henrik Woffinden ___ --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] Reception Console
Yes, please. I would love to test for you. Med venlig hilsen / Best regards, Henrik Woffinden Technical Director Nitram Lexa ApS Maglebjergvej 5A DK-2800 Kongens Lyngby Denmark Phone: +45 70 25 24 23 Fax: +45 70 25 29 23 Mobile: +45 40 85 25 17 E-mail: [EMAIL PROTECTED] Web: www.nitramlexa.com --- Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bit operating system originally coded for a 4-bit microprocessor by a 2-bit company that can't stand 1 bit of competition. Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales ___ --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: ZapHFC quadBRI D-Channel going down randomly
I have the exact same problem on a normal ISDN2 BRI line. I solved it by having my Telco put layer 1 to permanent. Best regards, Henrik Woffinden Alberto Pastore wrote: asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: 16 October 2006 17:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ZapHFC quadBRI D-Channel going down randomly Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point, others are point-to-multipoint. I keep getting always the same problem: after some hours of regular working, some boxes report the usual message Primary D-Channel on span n down (where n is different every time, depending on the number of active bri spans) I've read on previous postings that having layer 1 down on ptmp spans is normal. However after getting a down message (on ptp spans too!) I'm no more able to place outgoing calls on that span, until I restart asterisk zaptel drivers. Sometimes, they get back working by themselves (with the related span up notification) after a random time period. During the down period, incoming calls are regularly served. However these calls do not change the status of the span, i.e. as soon as the calls are hung up, the span gets down again. I've tried to capture the dialog between the card and NT1 equipment, and during the down state, I got this repeated over and over: Sending Set Asynchronous Balanced Mode Extended [ 00 8b 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 069EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] == Primary D-Channel on span 1 down In zapata.conf I'm pretty sure I've always set the correct signalling settings (switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending on the case) In /etc/zaptel.conf, I've tried many combinations with no difference; my current settings are like this: span=1,1,0,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,ami bchan=4-5 dchan=6 etc Any clue? Thanks, Alberto -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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: R: [asterisk-users] Zaphfc woth florz patch
Hi, You haven't applied the florz patch correct. You apply it like this : 1. Go to your bristuff directory 2.zcat [path to florz patch file]/zaphfc_0.3.0-PRE-1o_florz-12.diff.gz | patch -p1 When it is applied the you will get this output: [EMAIL PROTECTED] ~]# modinfo zaphfc filename: /lib/modules/2.6.17-1.2187_FC5.vs2.0.2.3/misc/zaphfc.ko description:HFC-S PCI A Zaptel Driver author: Klaus-Peter Junghanns [EMAIL PROTECTED] license:GPL vermagic: 2.6.17-1.2187_FC5.vs2.0.2.3 mod_unload 686 REGPARM 4KSTACKS gcc-4.1 depends: srcversion: 4FA4AFA6BDEBDA5885D440A parm: jitterbuffer:int parm: timer_card:int parm: sync_slave:int parm: debug:int parm: modes:int Note the 2 new parm Med venlig hilsen / Best regards, Henrik Woffinden Giordano Grandis wrote: First all i wrote syns_slave instead sync_slave, anyway also with sync_slave i got the same error. This is the modinfo output : modinfo zaphfc filename:/lib/modules/2.4.31/misc/zaphfc.o description: HFC-S PCI A Zaptel Driver author: Klaus-Peter Junghanns [EMAIL PROTECTED] license: GPL parm:modes int parm:debug int ...i have again the kpj ? Thanks again Giordano -Messaggio originale- Da: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Inviato: martedì 3 ottobre 2006 15.43 A: Giordano Grandis Oggetto: Re: [asterisk-users] Zaphfc woth florz patch On Tue, Oct 03, 2006 at 03:00:11PM +0200, Giordano Grandis wrote: Hi guys, i just installed the flortz patch with bristuff-0.2.0-RC8r but when i load module zaphfc i get this warning message: modinfo zaphfc ? Warning: ignoring syns_slave=0, no such parameter in this module Warning: ignoring timer_card=1, no such parameter in this module Module zaphfc loaded, with warnings Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) 6 channels configured. Asterisk starts without problem. Any ideas ? Thanks in advance Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Compile error in Asterisk 1.2.12.1
Hi. I'm using zaptel-1.2.9.1/libpri-1.2.3/asterisk-1.2.12.1 all patched with bristuff-0.3.0-PRE1s. What could be the problem when I get this compiler error: -- cut --- gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_agi.o res_agi.c res_agi.c: In function 'agi_exec_full': res_agi.c:2120: error: too few arguments to function 'launch_script' res_agi.c:2124: error: 'AGI' has no member named 'audio' res_agi.c:2094: warning: unused variable 'efd2' make[1]: *** [res_agi.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.12.1/res' make: *** [subdirs] Error 1 -- cut --- -- Med venlig hilsen / Best regards, Henrik Woffinden Technical Director Nitram Lexa ApS Maglebjergvej 5A DK-2800 Kongens Lyngby Denmark Phone: +45 70 25 24 23 Fax: +45 70 25 29 23 Mobile: +45 40 85 25 17 E-mail: [EMAIL PROTECTED] Web: www.nitramlexa.com --- Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bit operating system originally coded for a 4-bit microprocessor by a 2-bit company that can't stand 1 bit of competition. ___ --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] mISDN versus ZapHFC with BRIstuff
I have 2 single BRI s0 cards. -1 in TE mode for the outside line -1 in NT mode for the inside phones If I dial the group with Dial(Zap/g2/,60,t) then all MSN's on all phones ring. But how do I dial so only MSN 10,11,12 rings? If I dial every number as Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,t) then only 10 and 11 rings on separate b-channels and 12 is busy/congested. I know it can be done, cause my hardware PBX (Elmeg 46e) can do it using only 1 b-channel or through the d-channel. Best regards, Henrik Woffinden Kai Ober wrote: is it a single s0 card? how do you ring the 3 phones? no problems with the installation of mISDN so far. it is as easy as on Bristuff regards KAI Henrik Woffinden schrieb: Hi Sorry... I haven't been specific enough... I have several ISDN phones on my inside NT mode ISDN card, and I wan't 3 of the MSN (local) numbers to ring at the same time. I can't get more than 2 phones to ring at the same time, unless I ring them all by dialing the group, but that's not what I want. ___ --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] mISDN versus ZapHFC with BRIstuff
Hi Right now I'm running Asterisk with ZapHFC BRIstuff and it work, but with some MSN addressing problems on the ISDN bus. I've had no success solving the problem. Problem is making 3 MSNs ring on one B-channel. I thought of trying mISDN instead. Do I still need zaptel and libpri when using mISDN, or can I skip them totally? I thought maybe I needed zaptel for timing purposes? -- Best regards, Henrik Woffinden ___ --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] mISDN versus ZapHFC with BRIstuff
Hi Sorry... I haven't been specific enough... I have several ISDN phones on my inside NT mode ISDN card, and I wan't 3 of the MSN (local) numbers to ring at the same time. I can't get more than 2 phones to ring at the same time, unless I ring them all by dialing the group, but that's not what I want. The calls come in perfectly on my outside TE mode ISDN card. Best regards, Henrik Woffinden Remco Barendse wrote: On Thu, 14 Sep 2006, Henrik Woffinden wrote: Hi Right now I'm running Asterisk with ZapHFC BRIstuff and it work, but with some MSN addressing problems on the ISDN bus. I've had no success solving the problem. Problem is making 3 MSNs ring on one B-channel. I thought of trying mISDN instead. Do I still need zaptel and libpri when using mISDN, or can I skip them totally? I thought maybe I needed zaptel for timing purposes? I haven't tried MISDN yet, the installation looks much more complicated than bristuff but to answer your first problem... Are you sure that your telco is passing calls on all 3 MSN's? Bristuff doesn't decide which MSN rings on which B channel. It just detects and handles accordingly. If you never see a call coming in on the console for that MSN I would check with your telco first. Increase the verbosity of asterisk to find our what is going on. ___ --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] No dialtone, just directly busy
Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? -- Best regards, Henrik Woffinden ___ --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] No dialtone, just directly busy
That's exactly what happens: When I pick up the handle, this is what I get: -- Extension 's' in context 'from-inside' from '11' does not exist. Rejecting call on channel 0/2, span 2 Do you know what to do in the dialplan? Best regards, Henrik Woffinden Tim St. Pierre wrote: Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? ___ --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] No dialtone, just directly busy
immediate is already set to immediate=no, so that's not it. Best regards, Henrik Woffinden Eric ManxPower Wieling wrote: Remove immediate=yes from /etc/asterisk/zapata.conf Henrik Woffinden wrote: That's exactly what happens: When I pick up the handle, this is what I get: -- Extension 's' in context 'from-inside' from '11' does not exist. Rejecting call on channel 0/2, span 2 Do you know what to do in the dialplan? Best regards, Henrik Woffinden Tim St. Pierre wrote: Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? ___ --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] vim syntax highlighting( for Asterisk.conf files)
Hi I've got the exact same problem on Fedora Core 5 with vim70. Best regards, Henrik Woffinden Marco Mouta wrote: Hi all, I've just installed vim70, looking for vim syntax highlighting( for Asterisk.conf files) , http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting, and i notice that both: asterisk.vim and filetype.vim already refer asterisk configurations. But unfortunately i couldn't get yet the highlight syntax working fine for my asterisk.conf files. Any one can help me? Centos4.2 is my distribuition -- Com os melhores cumprimentos, Marco Mouta ___ --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 do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hello, Nobody has replied on this message. Isn't there anybody that has any input? Best regards, Henrik Woffinden Henrik Woffinden wrote: Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ascom Eurit 133 cordless ISDN phone
Hi, I have an Ascom Eurit 133 ISDN base station with 2 cordless handsets. I can receive calls excellent on these phones, but when I dial out Asterisk can't see what number I want to dial, and it routes me to the s extension. That rather unlucky for an outgoing call not to know the number you want to dial. If I put the cable directly in the NT box it works fine. I have 2 other kind of ISDN phones, and the work fine out through the same Asterisk. Anyone know what could be the trouble here? ISDN hardware in the Asterisk box is 2 ZAPHFC cards (1 in TE mode, and 1 in NT mode). Asterisk 1.2.10-BRIstuffed 0.3.0-PRE-1s -- Best regards, Henrik Woffinden ___ --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 do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hi, I've just tested that... And no, nothing on the channel rings. Henrik Woffinden Martin Polainer wrote: Hi, I have not tested yet, but maybe Dial(Zap/g1) would work; Guess this would ring everthing on Group 1... Best regards, Martin Polainer Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden: Hello, Nobody has replied on this message. Isn't there anybody that has any input? Best regards, Henrik Woffinden Henrik Woffinden wrote: Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot dial out through SIP provider
Hi again,
It was one letter which was wrong case in my secret
Sorry to have bothered with that problem.
Med venlig hilsen / Best regards,
Henrik Woffinden
Dovid Bender wrote:
- Original Message - From: Henrik Woffinden [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, August 27, 2006 11:50 AM
Subject: [asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=
fromuser=
secret=xx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[]
type=friend
context=internal
username=
secret=
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid=Henrik Woffinden
nat=yes
qualify=yes
insecure=very
;[EMAIL PROTECTED]
extensions.conf:
[internal]
;exten = _,1,Dial(Zap/g1/${EXTEN},,)
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
exten = _,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is and the number I dial is - which is a normal
mobile):
-- Registered SIP '' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial(SIP/-09f2eb28, SIP/[EMAIL PROTECTED]||) in
new stack
-- Called [EMAIL PROTECTED]
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to 'Henrik Woffinden
sip:[EMAIL PROTECTED];tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/-09f2eb28, ) in new stack
== Spawn extension (internal, , 2) exited non-zero on
'SIP/-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
Your sip provider is rejecting the call. This can be for many reasons.
Bad user/id pass, no credit left on acct., not using proper syntax
etc. Look at thier site and see how they want you to send the call to
them (i.e.with the + sign before the number or maybe add or remove a 0)
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[asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=
fromuser=
secret=xx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[]
type=friend
context=internal
username=
secret=
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid=Henrik Woffinden
nat=yes
qualify=yes
insecure=very
;[EMAIL PROTECTED]
extensions.conf:
[internal]
;exten = _,1,Dial(Zap/g1/${EXTEN},,)
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
exten = _,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is and the number I dial is - which is a normal
mobile):
-- Registered SIP '' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial(SIP/-09f2eb28, SIP/[EMAIL PROTECTED]||) in new stack
-- Called [EMAIL PROTECTED]
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to 'Henrik Woffinden
sip:[EMAIL PROTECTED];tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/-09f2eb28, ) in new stack
== Spawn extension (internal, , 2) exited non-zero on
'SIP/-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
--
Med venlig hilsen / Best regards,
Henrik Woffinden
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[asterisk-users] How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
Hello, I'm fairly new to Asterisk. Installation went fine, and things seem to work, but I have 1 problem. Hardware: 2 HFC ISDN cards (1 in TE mode and 1 in NT mode) 1 SIP On the inside (NT mode card) I have 3 ISDN phones. Everything is connected with all cables and extra resistors, and all 3 phones can dial and be dialled. When I try to dial all 3 phones simultaniously, with Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,m(default)t) then 2 phones ring and the last one is busy/congestion. I assume its cause I only have 2 b-channels. How do I make all 3 phones ring using only 1 channel? It can be done. I also have a hardware PBX (Elmeg C46) which does that now. Can anyone help me how to do it in Asterisk? -- Best regards, Henrik Woffinden ___ --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] pri rdnis found as Facility but not set
Hi, I'm running asterisk with a PRI. But I can't get hold of the rdnis number. When running pri debug I can see the true rdnis number as Facility, the number 703289840 as shown below. Is it possible to get hold of this value in some way from extensions.conf? Or is it necessary to modify the source for asterisk, in that case does someone know where and how? Thanks in advance, Henrik Protocol Discriminator: Q.931 (8) len=79 Call Ref: len= 2 (reference 40/0x28) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1c 26 91 a1 23 02 02 00 80 02 01 0f 30 1a 02 01 01 0a 01 02 a1 12 a0 10 a1 0e 0a 01 02 12 09 37 30 33 32 38 39 38 34 30] Facility (len=40, codeset=0) [ 0x91, 0xa1, 0x23, 0x02, 0x02, 0x00, 0x80, 0x02, 0x01, 0x0f, '0', 0x1a, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x02, 0xa1, 0x12, 0xa0, 0x10, 0xa1, 0x0e, 0x0a, 0x01, 0x02, 0x12, 0x09, '703289840' ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0a 21 83 31 38 31 33 34 32 35 35] Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '18134255' ] [70 05 c1 38 35 35 36] Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8556' ] -- Making new call for cr 40 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Aug 17 16:36:39 WARNING[31243]: chan_zap.c:8379 pri_dchannel: PRI_EVENT_RING Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] -- Accepting call from '18134255' to '8556' on channel 0/10, span 1 -- Executing Answer(Zap/10-1, ) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing NoOp(Zap/10-1, name: ) in new stack -- Executing NoOp(Zap/10-1, number: 18134255) in new stack -- Executing NoOp(Zap/10-1, ani: 18134255) in new stack -- Executing NoOp(Zap/10-1, dnid: 8556) in new stack -- Executing NoOp(Zap/10-1, rdnis: ) in new stack -- Executing Goto(Zap/10-1, test|1) in new stack -- Goto (default,test,1) -- Executing Answer(Zap/10-1, ) in new stack -- Executing Wait(Zap/10-1, 1) in new stack ___ --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] DTMF codes in feature.conf not comming through
I'm running Asterisk 1.2.7.1 using entirely SIP connections, but I have a problem with DTMF signaling. In the features.conf, I have set up sequences using * and # followed by a single digit for transfers etc. But when I then press '*' or '#' during a call, only each other is passed on. All other DTMF signals are working great. Is there a way to guarantee that single '*' and '#' are passed on (respecting a featuredigittimeout )? And is there a way do make a call NOT using the featuremap and therefore grapping the DTMF tones? On a call-to-call basis or for a specific SIP client? Regards, Henrik My feature map: [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 20 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 8 ; Number of seconds to wait between digits when transfering a call featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *3 ; One Touch Record atxfer = *7 ; Attended transfer ___ --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 setup a test number to know my extensionnumber
Hi,
In [EMAIL PROTECTED] it's done like this:
exten = *65,1,Answer
exten = *65,2,AGI(festival-script.pl|Your phone number is ${CALLERIDNUM}.)
exten = *65,3,Hangup
(extensions.conf or an include file)
You need to have Festival installed
/HZ
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 14, 2005 4:00 PM
Subject: [Asterisk-Users] How to setup a test number to know my
extensionnumber
I would like to setup a test number, that speaks back my phone number.
How can I set this up?
bye
Ronald
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[Asterisk-Users] Overriding SIP From Header
Is there a way to override the SIP From Header that is used in the extension.conf Dial command? The default is [EMAIL PROTECTED]. I do not want to configure SIP accounts in sip.conf, but instead generate the SIP From-User within extensions.conf from data the user has entered interactively. Any idea? Henrik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Overriding SIP From Header
Thanks, I know this. But is there a way to set these dynamically from within the Dialplan? Regards, Henrik -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Benjamin on Asterisk Mailing Lists Gesendet: Sonntag, 12. September 2004 21:50 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Overriding SIP From Header Is there a way to override the SIP From Header use fromuser= and fromdomain= in your peer entry in sip.conf rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P in Germany
Is there anyone successfully using the TE410P with a German PMX-Anschluss? Please just drop me a note mentioning the carrier you use. We are having problems making the card work, although configuration is correct (Posted this before). Our carrier blames the card for this. We would just need some evidence that it really works. Thanks, Henrik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard TE410P still making trouble
We are still having problems getting a Wildcard to work with a German E1 (PMX) interface. When starting asterisk it shows all B-channels starting up successfully (although our carrier told us only the first B-channel starts, if any at all). Incoming calls are not being signaled at all. (They seem to be intercepted by the carrier's switch, as no B-channel is up) Outgoing calls sometimes work, but only on B-channel 1. Anyone has seen this problem before? Digium support was not able to solve this, so we are kind of stuck. Henrik -- Call log for succesfull call on B-channel 1: -- Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] [70 0c c1 30 38 30 30 38 30 38 30 38 30 30] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '08008080800' ] [a1] Sending Complete (len= 1) -- Called g1/08008080800 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32770/0x8002) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32770/0x8002) (Terminator) Message type: PROGRESS (3) [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [1e 02 82 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 32770/0x8002) (Terminator) Message type: CONNECT (7) [29 05 04 09 06 0c 1c] Time Date (len= 7) [ 04-09-06 12:28 ] -- Processing IE 41 (cs0, Date/Time) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/1-1 answered SIP/sipsnip.com-081b7340 -- This is what we get on other channels: -- Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] [70 0c c1 30 38 30 30 38 30 38 30 38 30 30] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '08008080800' ] [a1] Sending Complete (len= 1) -- Called g1/08008080800 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32774/0x8006) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 ac] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (2) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/2, span 1 got hangup -- Forcing restart of channel 0/2 on span 1 since
[Asterisk-Users] WG: Digum TE410P
Hello, We recently installed a Wildcard TE410P, but we are having problems to make it work reliably with a German E1 (Primaermultiplexanschluss PMX DSS1). Our carrier (Hansenet in Hamburg, Germany) is using Nokia/Lucent switches. The card is only able to set up the first B-channel (although it tells via Asterisk, that all B-Channels have been successfully started). When connecting a testing device to the E1 it is able to get all B-channels. When switching the cord to the TE410P the card keeps the B-channels, but releases them after the first incoming call (this is what our carrier told us). Sometimes it keeps a single channel long enough for us to make some calls. Is anyone experiencing the same problem? Maybe our configuration files are incorrect? Can anyone help? Thank you, Henrik Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3,crc4 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 loadzone=de defaultzone=de Zapata.conf [channels] language=en context=default switchtype=euroisdn pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-15 switchtype = euroisdn signalling = pri_cpe group = 1 channel = 17-31 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 32-46 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 48-62 switchtype = euroisdn signalling = pri_cpe group = 3 channel = 63-77 switchtype = euroisdn signalling = pri_cpe group = 3 channel = 79-93 switchtype = euroisdn signalling = pri_cpe group = 4 channel = 94-108 switchtype = euroisdn signalling = pri_cpe group = 4 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] WG: Digum TE410P
Thanks, I made the changes, but that did not help?! This is what Asterisk is telling me when I try to dial out. Incoming calls are not signaled at all. When our carrier resets the E1, the first channel sometimes works for a short time: *CLI -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 -- B-channel 0/24 successfully restarted on span 1 -- B-channel 0/25 successfully restarted on span 1 -- B-channel 0/26 successfully restarted on span 1 -- B-channel 0/27 successfully restarted on span 1 -- B-channel 0/28 successfully restarted on span 1 -- B-channel 0/29 successfully restarted on span 1 -- B-channel 0/30 successfully restarted on span 1 -- B-channel 0/31 successfully restarted on span 1 -- Executing Dial(SIP/sipsnip.com-081b3798, Zap/g1/c04044x) in new stack -- Called g1/c04044506451 -- Channel 0/1, span 1 got hangup -- Forcing restart of channel 0/1 on span 1 since channel reported in use -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(SIP/sipsnip.com-081b3798, ) in new stack == Spawn extension (default, , 2) exited non-zero on 'SIP/sipsnip.com-081b3798' -- B-channel 0/1 successfully restarted on span 1 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Michael Bielicki Gesendet: Donnerstag, 2. September 2004 23:31 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: Digum TE410P You cannot take the primary clock from all 4 channels. So you must change your lines in zaptel.conf to: span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 span=3,3,0,ccs,hdb3,crc4 span=4,4,0,ccs,hdb3,crc4 also why specify the groups double ? switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15,17-31 cheers Michael On Thu, 2004-09-02 at 23:16, Henrik Pfluger wrote: Hello, We recently installed a Wildcard TE410P, but we are having problems to make it work reliably with a German E1 (Primaermultiplexanschluss PMX DSS1). Our carrier (Hansenet in Hamburg, Germany) is using Nokia/Lucent switches. The card is only able to set up the first B-channel (although it tells via Asterisk, that all B-Channels have been successfully started). When connecting a testing device to the E1 it is able to get all B-channels. When switching the cord to the TE410P the card keeps the B-channels, but releases them after the first incoming call (this is what our carrier told us). Sometimes it keeps a single channel long enough for us to make some calls. Is anyone experiencing the same problem? Maybe our configuration files are incorrect? Can anyone help? Thank you, Henrik Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3,crc4 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 loadzone=de defaultzone=de Zapata.conf [channels] language=en context=default switchtype=euroisdn pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-15 switchtype = euroisdn signalling = pri_cpe group = 1 channel = 17-31 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 32-46 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 48-62 switchtype = euroisdn signalling
AW: AW: [Asterisk-Users] WG: Digum TE410P
Yes, they told me so. When I turn it off I only get endless messages: ep 3 02:28:37 WARNING[98310]: chan_zap.c:1880 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 up Sep 3 02:28:38 WARNING[98310]: chan_zap.c:1880 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 up Sep 3 02:28:39 WARNING[98310]: chan_zap.c:1880 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Michael Bielicki Gesendet: Freitag, 3. September 2004 00:13 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] WG: Digum TE410P Hmm I have seen that behaviour before. Are you sure hansenet uses crc4 ? On Fri, 2004-09-03 at 00:00, Henrik Pfluger wrote: Thanks, I made the changes, but that did not help?! This is what Asterisk is telling me when I try to dial out. Incoming calls are not signaled at all. When our carrier resets the E1, the first channel sometimes works for a short time: *CLI -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 -- B-channel 0/24 successfully restarted on span 1 -- B-channel 0/25 successfully restarted on span 1 -- B-channel 0/26 successfully restarted on span 1 -- B-channel 0/27 successfully restarted on span 1 -- B-channel 0/28 successfully restarted on span 1 -- B-channel 0/29 successfully restarted on span 1 -- B-channel 0/30 successfully restarted on span 1 -- B-channel 0/31 successfully restarted on span 1 -- Executing Dial(SIP/sipsnip.com-081b3798, Zap/g1/c04044x) in new stack -- Called g1/c04044506451 -- Channel 0/1, span 1 got hangup -- Forcing restart of channel 0/1 on span 1 since channel reported in use -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Hangup(SIP/sipsnip.com-081b3798, ) in new stack == Spawn extension (default, , 2) exited non-zero on 'SIP/sipsnip.com-081b3798' -- B-channel 0/1 successfully restarted on span 1 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Michael Bielicki Gesendet: Donnerstag, 2. September 2004 23:31 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: Digum TE410P You cannot take the primary clock from all 4 channels. So you must change your lines in zaptel.conf to: span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 span=3,3,0,ccs,hdb3,crc4 span=4,4,0,ccs,hdb3,crc4 also why specify the groups double ? switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15,17-31 cheers Michael On Thu, 2004-09-02 at 23:16, Henrik Pfluger wrote: Hello, We recently installed a Wildcard TE410P, but we are having problems to make it work reliably with a German E1 (Primaermultiplexanschluss PMX DSS1). Our carrier (Hansenet in Hamburg, Germany) is using Nokia/Lucent switches. The card is only able to set up the first B-channel (although it tells via Asterisk, that all B-Channels have been successfully started). When connecting a testing device to the E1 it is able to get all B- channels. When switching the cord to the TE410P the card keeps the B-channels, but releases them after the first incoming call (this is what our carrier told us). Sometimes it keeps a single channel long enough for us to make some calls. Is anyone experiencing the same problem? Maybe our
[Asterisk-Users] Re: Call queues
Hi Jeremy, What about this in extentions.conf exten = 5000,1,Dial(SIP/phone1SIP/phone2SIP/phone3,50,r) -- mvh. Hans-Henrik Andresen Jeremy Kenney [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help -Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem
Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem
hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- usedcanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MYSQL_FRIENDS and IAX problem
Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zapras - and kernel ??
Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel Plugin Initialized Jul 15 11:43:57 voip1 pppd[9299]: Using zaptel device 'stdin' Jul 15 11:43:57 voip1 pppd[9299]: pppd 2.4.1b2 started by root, uid 0 Jul 15 11:43:57 voip1 pppd[9299]: Zaptel device is 'stdin' Jul 15 11:43:57 voip1 pppd[9299]: Unable to put device 'stdin' into HDLC mode Jul 15 02:43:57 voip1 kernel: Zaptel: Zaptel PPP support not compiled in Jul 15 11:43:57 voip1 pppd[9299]: Exit. It's thrue, I have'nt pathced and compiled the kernel, but Can't find anything about it - no reademe. Any clue ? or any how-to for the zapras ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on 64bit ?
Hi, A'm about to set up a asterisk for 5000 users, and the customer had a 64bit server - can asterisk compile on that ? I will use a digium X100P for timing use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6) What else ? Is it posible to have only one server for 5000 users ? I gues that it will be 5-700 sim. users only talking sip, and IAX2 to my PSTN-Gateway. The system is suposed to scale to 15000 users. I'm ready to receive input :) /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk on 64bit ?
Hi, Thanks for your answer: Yes it works, in theory... (You *might* run into some hardcoded limits but they are usually easy to fix) I was just wondering if someone HAS it running, could tell me it WONT work at all. It'll be really interesting to see 15000 registrations on one server... Sory, misunderstanding, no - I was thinking on 5000 users per server Just a small piece of advice... try using more and (physically) distributed servers. I will do There are basicly 1000 ways to solve this and each member of this list Yea. I read on the wiki-pages that it was not good to use agi, cause of heavy load on the server, but an extensions.conf with 5000+ entrys is that good ? I would preffer agi and a very littlte and simple extensions.conf. Any experience with asterisk and 5000-15000 users ? -- mvh. Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling festival
Hi, I had followed the installation-guide to festival http://www.voip-info.org/wiki-Asterisk+festival+installation speech-tools compiles OK, but I got this error when compiling asterisk if I compile without the patch it compiles, but of cause did'nt work with asterisk. any clue ? /Hans-Henrik Andresen Making in directory src/modules/base ... making dependencies -- modules.cc module_support.cc parameters.cc ff.cc pos.cc p hrasify.cc word.cc postlex.cc phrinfo.cc g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include modules .cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/festival.h:44, from modules.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include module_ support.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/module_support.h:45, from module_support.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include paramet ers.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/module_support.h:45, from parameters.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include ff.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/festival.h:44, from ff.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include pos.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from
[Asterisk-Users] 'Answered' at wrong time.
Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Whats wrong ? Can I do something about it ? /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] notransfer=yes but still tryin to bridged
Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 'Answered' at wrong time.
Problem at partner site, some perl-problem with answer-command /HHA Whats wrong ? Can I do something about it ? /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP response 404 Not Found AND circuit-busy ??
I have a dlink dvg-1120s voip-router. I can make calls out from the router, but when calling the router I got -- Executing Dial(SIP/2010-b437, SIP/2021|30|r) in new stack -- Called 2021 -- Got SIP response 404 Not Found back from 62.79.78.74 -- SIP/2021-473b is circuit-busy What does this meen ? Or what can I do ? The router is behind nat, but if I put the router on the same network as asterisk it work ok /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 reload - how ?
Hi, My asterisk fails and stops after running the reload command ~20 times (I'm testing) - is this a kown problem ? Therefor I wil reload only sip, extensions and iax, it works with sip and extensions, but it seem that there are no reload for iax - or what ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Newbie....
If you want MusicOnHold and Conferencing however, you will need one card for the timing. Why - I had used only ztdummy that works for MOH and conf. uncomment ztdummy in the makefile for zaptel and compile. /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connection to Cisco Call Manager
Hi, At my company we have a large CCM-installation, is it possible to / how to connect between asterisk and CCM. I'm quit shure that the CCM only use Skinny. Any idea of the hardware-size for 1000 users ? /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Re: Limit on call in minuttes.
Tanks This was exatly what I needed, /Hans-Henrik Andresen Nicolas Gudino [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Hans, http://bugs.digium.com/bug_view_page.php?bug_id=773 This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit on call in minuttes.
Hi, I saw somewhere that it was possible to set a limit for how long time a call could be, for an extension in extension.conf. But I can't find it anymore. Can someone please help. Calls to '411' an operator may max. be 5 min. I have this in extension.conf. [shortcuts] exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) [operator] exten = 0,1,Dial(SIP/operator,30,tr) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Help Newbie: TDM Development Kit
Did you compile the zap and lipri and installed ? /HHA app_dial.c:533 dial_exec: Unable to create channel of type 'ZAP' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] peer is UNREACHABLE when using XLITE
Hi, I have 3 friends trying to connect to my Asterisk using x-lite, all of them are using 3 dif. adsl-provider. For each of them I got this in sip.conf: disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=g723.1 [seholm] type=friend secret=** auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=inband callerid=Svend Erik Holm 60 context=sip They can all connect, but in asterisk I got this *CLI Mar 7 09:20:26 NOTICE[278546]: chan_sip.c:5846 sip_poke_noanswer: Peer 'henrikoglone' is now UNREACHABLE! And sip show peers show this seholm 83.88.89.122(D) 255.255.255.255 5060 UNREACHABLE They can make calls TO me, but of cause the pickup wont be sent to them, so asterisk shut down the channel after 5 sec. or so. If they are using sjphone it works, test with gs. analog adaptor also works. If I use a x-lite on same lan as asterisk, and if they make a VPN to my asterisk it works. (We had tried to use stun-server as well) Any clue ? /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
I have no problem transfer from one GS adaptor to another GS adaptor. /Hans-Henrik Andresen Can anyone confirm that this problem exists? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: peer is UNREACHABLE when using XLITE
Hi I tried to raise it to 5000, but still unreachable. But as I wrote earlyer, for the same config, sjphone and a Grandstream 286 works. /HHA qualify=1000 If the client turns UNREACHABLE, you might want to change the qualify= setting to qualify=yes, that defaults to two seconds, instead of one second that you have here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Limit on call in minuttes.
Hi, Thank you, but this I cant get to work. /HHA so that should enable you to do the following: Call timeout = 20 sec Max Call Duration = 300 sec = 5 min. exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300)) however, I have not tried it yet so someone correct me if I am wrong.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Limit on call in minuttes.
Thank you This works, but. It just cut the line, I had hoped for some bip bip bip to remind that now your about to be disconected, is this possible as well ? /Hans-Henrik Senad Jordanovic [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] exten = 1,AbsoluteTimeout ($SECONDS) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Limit on call in minuttes.
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
Soren Rathje [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
From extensions.conf:
[pstn-out-nat]
;
ignorepat = 0
; NOT USED
exten = _0XX0X,1,Congestion
; Local eight-digit dialing accessed through trunk interface
exten = _0NXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},20,S(20))
exten = _0NXXX,2,Congestion
From * console:
-- Executing Dial(SIP/1000-4d25, Zap/1/4060|20|S(30)) in new
stack
-- Setting call duration limit to 30 seconds.
-- Called 1/4060
-- Zap/1-1 answered SIP/1000-4d25
-- Hungup 'Zap/1-1'
== Spawn extension (default, 04060, 1) exited non-zero on
'SIP/1000-4d25'
cdr_odbc: Query Successful!
-- Søren
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec = 5 min.
exten =
411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300))
however, I have not tried it yet so someone correct me if I am wrong..
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[Asterisk-Users] Re: Re: Re: Limit on call in minuttes.
arhh - I did a checkout the 4th of marts - I will do a new checkout /HHA When did you checkout your version of Asterisk from CVS ?? This feature was put into CVS on the 6'th as a fix for bug #1107 but I have not seen it in v1-0_stable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Re: Limit on call in minuttes.
What checkout name should I do ? Just asterisk ? # cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot# cvs login - the password is anoncvs.# cvs checkout asterisk /HHA This is a new feature, that's why it is NOT in 1.0-stable. Only bugfixes go into -stable. New feautres - in CVS. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] danish voice
Hi, Anyone got danish voice-files who wants to share ? /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling zaptel
Hi, On my Suse90-out of box I had downloaded from CVS asterisk. I'm running kernel 2.4.21-99-smp4g with 4cpu's, and the kernelsource is in /usr/src/linux Asterisk compiles with no problem. But when compiling zaptel I got this error .. zaptel.c: In function `zt_ec_chunk': zaptel.c:4981: error: parse error before unsigned zaptel.c: In function `process_timers': zaptel.c:5500: error: parse error before unsigned zaptel.c: In function `zt_timer_poll': zaptel.c:5513: error: parse error before unsigned zaptel.c: In function `zt_chan_poll': zaptel.c:5545: error: parse error before unsigned zaptel.c: In function `zt_transmit': zaptel.c:5728: error: parse error before unsigned zaptel.c: In function `zt_receive': zaptel.c:5820: error: parse error before unsigned zaptel.c:5842: error: parse error before unsigned zaptel.c:5852: error: parse error before unsigned zaptel.c:5878: error: parse error before unsigned zaptel.c:5889: error: parse error before unsigned zaptel.c:5892: error: parse error before unsigned make: *** [zaptel.o] Error 1 Any help on this ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Error compiling zaptel
Greate - /usr/src/linux/include/linux/version.h:6:2: #error The kernel sources in /usr/src/linux are not yet configured. /usr/src/linux/include/linux/version.h:7:2: #error Please run 'make cloneconfig make dep' in /usr/src/linux/ /usr/src/linux/include/linux/version.h:8:2: #error to get a kernel that is configured like the running kernel. /usr/src/linux/include/linux/version.h:9:2: #error Alternatively, you can copy one of the config files /usr/src/linux/include/linux/version.h:10:2: #error arch/$ARCH/defconfig.* to .config, and run /usr/src/linux/include/linux/version.h:11:2: #error 'make oldconfig make dep' to configure the kernel /usr/src/linux/include/linux/version.h:12:2: #error for that configuration. h - I done the 'make cloneconfig make dep' and do I have say that i works now - thank you. you left out the important part of the error log. I guess the compiler complains about a header file it doesn't find before throwing all those parse errors. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maillinglist as newsgroup ?
Hi, I was thinking if it was possible to get this list as news ? It would be much easier that 'hotmail-account' /HHA _ Scope out the new MSN Plus Internet Software optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Maillinglist as newsgroup ?
Greate - it works. Thank you /HHA http://www.gmane.org offers many mailinglists as a newsfeed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] configuration to Grandstream via tftp
Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like tftpserver-dir mac-address firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _ Rethink your business approach for the new year with the helpful tips here. http://special.msn.com/bcentral/prep04.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] configuration to Grandstream via tftp
Thanks. How is the directory structure ? or do you add all you phone to the one file cfg.txt and have it in the root of your tftp-dir ? /HHA Attached is the config file I send to my Grandstream. Change IP address Phone ID to suite. _ Find high-speed net deals comparison-shop your local providers here. https://broadband.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] configuration to Grandstream via tftp
Thank your for the link - now I wil try it :) /Hans-Henrik Andresen This is the URL I got the config file from, http://www.plugndial.com/ it's on a link from the SipPhone URL. _ Learn how to choose, serve, and enjoy wine at Wine @ MSN. http://wine.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GS Handytone Echo-problem
Hi, Yesterday I finaly got my handytone sip adaptor. It works But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _ Scope out the new MSN Plus Internet Software optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN30 - HW ?
Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _ Find high-speed net deals comparison-shop your local providers here. https://broadband.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN30 - HW ?
Yes, ISDN30 ( I am assuming you are in the UK) is basically an ISDN PRI line which is basically an E1 line, so you would need to get an E100P card from Digium to be able to connect your ISDN30 into Asterisk.. I'm from Denmark (else my english would had been better:( ) As for the rest of the questions I can't reallt answer as I have never personally connected an E100P to an ISDN30 line.. many on this list have and will hopefully be able to give you more of the technical details.. later.. I'll wait to see if some one else can help. (Wipeout - nothing about the echo ?) _ Check out the coupons and bargains on MSN Offers! http://shopping.msn.com/softcontent/softcontent.aspx?scmId=1418 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-Client for Handheld PC
Hi, Yes Telesym, xten and one more I can't remember the name of it, they are all for PPC-only. :( /HHA From: Ray Burkholder [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC Date: Mon, 12 Jan 2004 14:33:39 -0500 What are the ones you found for PocketPC? I guess you've looked at the Telesym site? They have a SIP flavor coming out shortly for some PDA's. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik Andresen Sent: January 12, 2004 05:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-Client for Handheld PC Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _ Let the new MSN Premium Internet Software make the most of your high-speed experience. http://join.msn.com/?pgmarket=en-uspage=byoa/premST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth ? + Doc + cdr
Hi, How much bandwidth do I need for 1 conversation ? I know it depends on the codecs, in X-lite I can see a codec called gsm, and the grandstream aha analog/ip converter have a codec called 721. Doc. I have found the asterisk handbook, but only a draft from marts 2003 anything newer ? Guides/howtos are welcome as well. anyone have a php interface to accounting ? /HHA _ There are now three new levels of MSN Hotmail Extra Storage! Learn more. http://join.msn.com/?pgmarket=en-uspage=hotmail/es2ST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
