Displaying 20 results from an estimated 11000 matches similar to: "No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4)"
2005 Oct 16
2
No voice - one way - both ways
I got four phones:
601 is a SIP phone (no brand)
615 is Snom 190
621 is a Grand stream
628 is a remote SIP phone (no brand)
601, 615, 628 can call each other without any problems
621 used to be able to call remote 628, but after upgrade to CVS Head
Nov. 11 the remote party cannot hear me.
615 never could call remote 628, both party hear nothing.
601 can always call 628
[Oct 16 00:52:13] --
2006 Jun 04
3
transfer & other features
*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * *0
Dial option is tTwWr
I tried to call from 601 to 615
601 keys in *0
2006 Jan 29
2
username not stabled?
vpbx*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
621/621 192.168.250.76 D N 5060 OK (65 ms)
626/626 192.168.250.109 D N 5060 OK (180 ms)
616/Ronald Softphone (Unspecified) D N 0 UNKNOWN
615/Ronald office 192.168.250.103 D N 5060 OK (41
2006 Jan 27
2
Name/username (sip show peers)
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
bye
Ronald Wiplinger
2004 Aug 03
6
features.conf
Is features.conf included in the cvs as of 8-1-04? I have updated, but am
not seeing it?
2004 Nov 29
1
Calling from PSTN let exension 601 ring twice, hang up and starts over again to ring twice, ...
Calling from PSTN let extension 601 ring twice, hang up and starts over
again to ring twice, ...
If I pickup I do not hear on extension 601, and on the PSTN it is still
signaling to ring.
Can anybody enlighten me, please?
extension.conf
[incoming_88097074]
exten => s,1,Wait(1) ;wait to get caller ID in.
exten => s,2,Dial(SIP/102,20)
exten => s,3,Voicemail(u102)
exten =>
2006 Apr 04
1
voipstunt: "Forbidden - wrong password ..."
voipstunt: "Forbidden - wrong password on authentication for INVITE to ...."
I have paid, the password was not changed, ... I have no idea why.
Is there anything what I can do to get this "failed" call over to
another provider, so that the user can complete the call?
(Dialstatus was an idea, but the line does not show up in CLI)
[Apr 5 09:22:36] -- Executing
2005 May 10
2
Stun & codec
I have two phones, one does not need stun, the other one needs.
All settings are identically, except the number/password and said above
stun - not stun
I use codec in the order:
g729
g711u
g711a
Any ideas, why the user can hear me, but I cannot hear him (stun) while
the other user without stun has no problem.
bye
Ronald
2005 Mar 06
3
SJphone on PDA registering with Asterisk???
I try to setup SJphone on my PDA, but it does not register with Asterisk.
I have setup a sip account on asterisk, ...
Can anybody give me a hint?
bye
Ronald
2005 Sep 02
2
Sipura 3000 setup
Can anybody show me a working Sipura 3000 setup please?
I need to setup one to my * box, ...
What are the variants you can setup? Advantage - disadvantage.
bye
Ronald Wiplinger
2010 Oct 01
2
No translator path exists for channel type DAHDI (native 76) to 256
Hello,
We are having issues with a NEW Sangoma A108D:
-- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0",
"DAHDI/g0/691918892|30|m") in new stack
[Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator
path exists for channel type DAHDI (native 76) to 256
[Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to
create
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers: 0.7269 or 0.2929 ???
bye
Ronald Wiplinger
2005 Mar 14
2
Has anybody experience with SetGroup / CheckGroup commands?
I am checking on the SetGroup / CheckGroup commands, but I have some
troubles to undestand the examples.
SetGroup(moh) can be moh anything as I like? Usually moh stands for
"music on hold"
CheckGroup(1) checks if somebody in in group "moh". Does it mean I can
only have one SetGroup(xxx) ??
When I look at example 2 than I see two SetGroup commands and one
CheckGroup
2003 Nov 21
4
Unable to create channel of type 'SIP'
I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
Executing Dial("Zap/1-1", "SIP/100|20") in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create channel of type 'SIP'
== Everyone is busy at this time
Has anyone seen this issue before?
2005 Aug 17
1
Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below
I go off hook
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
2006 Mar 31
4
How to check if a phone / line is used?
In the past I used SetGroup and CheckGroup to figure out if my allowed
providers lines are all used or not.
Since most of my provider have given me a single line anyway, I wonder
if there is a way to check if this (provider) line is taken already.
How can I do that?
Same is with the phone. How can I see in CLI if a phone is now in use or
not?
"Sip show peers" shows me just if it is
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2004 Jul 27
2
Enum
You can play also with www.enum2go.com <http://www.enum2go.com/> or
wap.enum2go.com
Regards
Alex
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040727/6c42c39d/attachment.htm
2005 May 26
5
SIP Soft Video phone for Asterisk usage
I am looking for a SIP Soft Video phone, which I can use with Asterisk.
If you have one installed (regardless if free or purchased) please tell
me which one, the settings in Asterisk and your experience with it.
bye
Ronald