Displaying 20 results from an estimated 5000 matches similar to: "SJphone Do not send silence - option ? Should be disabled for Asterisk"
2006 Mar 24
2
SIP trunk problem
Hi all,
I have the following problem, working with a SIP provider, if i setup my
SJPhone to register directly to their STUN server and working over a 384/128
ADSL i have a really good quality, but then if i configure Asterisk to
register to the same provider over the same 384/128 circuit the quality is
REALLY BAD. The obvious difference is that using directly the SJPhone i am
using STUN, while
2006 Mar 30
1
Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best regards,
Marco Mouta
2006 Mar 15
6
Cisco phones and Linksys SRW224P
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable!
Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot
2006 Mar 01
9
MOH native files
Where can I find alaw, ulaw, gsm, g729 formats for native music on hold?
I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3?
Thank you for your time!
--
Tomislav Parcina
tparcina#lama.hr
2006 Mar 07
2
Send One Touch Record to mail
How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf?
Thank you for your ideas.
--
Tomislav Parcina
tparcina#lama.hr
2003 Jun 17
3
sip.conf
HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register => user:password@host:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks a lot in advanced
michelle
-----
Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam
2006 Feb 09
4
Queue - check agent
I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call.
Thank you for your time.
--
Tomislav Par?ina
Lama Computers Split
2006 Feb 13
4
Voicemail - direct call
Hi list!
How to send a call directly to voicemail recording?
When I put this
exten => 313,n,VoiceMail,u221
Or this
exten => 313,n,VoiceMail,b221
In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible?
--
Tomislav Parcina
tparcina#lama.hr
2006 Feb 14
5
Call centre - * hang's up
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0.
I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help
2006 Feb 09
5
What ATA should I buy?
I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong.
Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this.
One more question, can I plug two lines in any of
2006 Apr 05
3
queue issue
Hi,
I have several queues configured at my call center for different support levels.
Today, something weird happened:
- A client called queue 1 and was answered by an agent
- The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf
- The user transferred the client to another Queue, by using the second channel and the XFer key of her
2006 Apr 18
2
Cisco 7970 SIP - few questions
- How to restart the phone? (On 7960 it is *+6+Settings)
- How to setup working dtmf?
- How to setup hinting?
For line is
<line button="4">
<featureID>9</featureID>
...
For speeddial is
<line button="5">
<featureID>2</featureID>
<featureLabel>341</featureLabel>
<speedDialNumber>341</speedDialNumber>
</line>
2006 Feb 27
5
res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind
2006 Mar 02
4
Info about F1000G
Does anybody use UTStarcom F1000G Wi-FI VoIP phone?
http://www.utstar.com/Solutions/Handsets/WiFi/
I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point?
Please, any information's are useful to me.
--
Tomislav Parcina
tparcina#lama.hr
2006 Oct 27
4
IAX2 show peers - description
Hi people,
pls does anybody know what "(T)" and "(D)" letter means?
server3*CLI> iax2 show peers
Name/Username Host Mask Port Status
SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK
(29 ms)
SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK
(95 ms)
2 iax2 peers [2 online, 0 offline, 0
2005 Dec 28
3
voip-info: Asterisk record calls
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there
is "Example by Mojo". I have done everything he said and I have sox
package installed.
[root@pbx recordings]# sox -help
sox: Version 12.17.7
...
When I open this web page http://10.0.0.26/recordings/index.php I get
this: No Recordings Found
And there are recordings in /var/spool/asterisk/monitor
Do I have to do
2003 Nov 01
13
Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I prefer not to go
back to RH7.3...
BTW, where would I find a useful FM?
David
--
David J. Sussman, MBA
email:
2006 Mar 16
7
OT: Unblocking bloced CID
Hello list, I know this has been brought up before but
I dont think there was ever a final answer. Is it
legal in the US to modify asterisk to show the CID
information that was received as blocked ? Thanks.
Dovid p.s. Sorry for the poor typing format, it was
written from a mobile phone.
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2006 Mar 28
3
Agent in multiple queues?
Hi,
What do I need to do to put an agent into two queues? The idea being
that the agent will get the call no matter which queue it comes into?
~ Matt
2006 Apr 27
2
Transfer - context/priority
Hi list!
When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"?
The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call