In your settings button 1stly unlock config.Then find SIP
Configuration (option 4) after selecting that go to option 7 that is
Messages URI. Now edit or put the extension.
Thank you
Yusuf
On 8/11/05, Lokesh kumar [EMAIL PROTECTED] wrote:
Hi,
Everybody
I am running asterisk successfully, I am
Another thing you can do. In your dialplan you can define cid in
incoming call like this..
exten = s,1,SetCallerID(Unavailable)
Thanks
yusuf
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I need the same for Cisco 7960 with sip image. I need to rearrange the
softkeys of the phone.
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I am replying to me!!! I am extremely sorry guys for asking that
stupid question!! My problem is solved. I connect that telular to the
fxo port of my pbx and nothing else was required.
Thank you all.
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feature is basically forward the call to another number(May
be my concept is still not clear!).
Person X might be at Y's desk,and wants to login Y's phone with his(X)
extension.So I can't use call forwarding.
Regards
Yusuf
On 7/19/05, C F [EMAIL PROTECTED] wrote:
On 7/19/05, Yusuf Iqbal [EMAIL
in a distant location
from my desk for a long time I could be able to get my phone with me.
Thank you,
Yusuf Iqbal
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I have a telecomFM CellRoute GSM. I want to route call to the cell
phones with that device. Has anyone experienced in connecting an
Asterisk pbx to telecomFM CellRoute GSM
successfully? Please help me to integrate this device with Asterisk Server.
Thanx in advance,
Yusuf Iqbal
Julien,
Thanks for your suggestions. You are working on it right? Could you
please inform us about your updates? So that we could get those
buttons working properly and console become clean!
On 6/2/05, Julien Goodwin [EMAIL PROTECTED] wrote:
On Thu, Jun 02, 2005 at 01:56:09PM +0600, Yusuf
7910's with Chan_sccp are running well except some cases.
As usually Transfer, Msgs, Conf,Forward,Speed buttons are not working!
Infact I have already knew that, those buttons will not work (from
mailing list).
It seems like the * console is very busy with messages constantly on
it from the sccp
Hi Andy,
Thank you so much for your help. My 7910's are now working!!!:) Now I
can work with those IP phones. I am still monitoring them. I will let
you know the further status.
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Hi Andy,
I have been trying to get 7910's work with *. I have tried with both
skinny and chan_sccp. Could you please instruct me about the
configuration? I have found some detalis about 7920 with sccp in
voip-info.org. But I haven't find any document for 7910. Please help
me to get them work
Scott Stingel wrote:
Some questions:
What country are you in?
Is there anything else connected to the line from the PSTN? It sounds
like you have a marginal condition, such as insufficient loop current
perhaps.
Do have any features, such as call waiting, on the line?
Do you know how far
I have bought some Wildcard X101P and Generic Clones for my Asterisk
PBX. Now I can place and get calls through the lines/channels.
Everything is okay but the problem is when I call outside through our
PSTN line, after few minutes the connection breaks down. The same
thing happens in case of
I have a couple of Cisco 7910's and I'd like to get them working with
asterisk. I have two X101P wild cards installed and they are
functioning well (Other two cards showing Red Alarm after few minutes
conversation).
I have configured foure cisco 7960 with SIP and they are working fine
with *.I
Can anybody help me to figure out how much memory per minute is consumed
for voicemail applications? And how many concurrent calls can be
handled at a time in Asterisk? so that, I can choose the specification
for Server to setup Asterisk for a large number of users.
Previously I have posted the same mail but no one answered me...Sorryfor resending the mail.I have bought a Wildcard X101P for my Asterisk PBX. Now I can placeand get calls through the lines/channel. Everything is okay but theproblem is when I call outside through our PSTN line, after fewminutes
I have boughta Wildcard X101P for my Asterisk PBX. Now I can place and get calls through the lines/channel. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of incoming calls. I have
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