Displaying 20 results from an estimated 4000 matches similar to: "Call forward on unavailable timer issues"
2004 Apr 12
2
SwissVoice IP10S not able to dial calls
I have set up a new SwissVoice phone and it can receive calls but I cannot
make calls out from it. The setup is simple for now, 2 phones: SwissVoice
is ext 7726 and Cisco 7960 (SIP) is ext 7999.
I can call from the Cisco phone and it rings on the SwissVoice phone but
when I dial from the SwissVoice phone I get a busy tone upon dialing the
second digit. The log reads as follows:
-- Endpoint
2004 Sep 28
0
Leader IP10S
Funny - I downloaded the latest Asterisk CVS, and it's pretty much working.
Will report when I have some more success.
PaulH
-----Original Message-----
From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de]
Sent: Tuesday, 28 September 2004 9:46 PM
To: Paul Hales
Subject: Re: [Asterisk-Users] Leader IP10S
Hi!
> I have been lent a Leader IP10S phone (SIP) for
2004 Jun 22
0
swissvoice ip10s firmware?
Hi,
Does anybody know the place to download the firmware for swissvoice ip10s
I have several phones with application IP10 H3 v1.0.0 (Build 1)
I'm looking for newer H.323 and also MGCP firmwares
Are the SIP firmware available, according to web its targeted to Q1 2004,
but we have week left in Q2
I sent several email to swissvoice support,, no answers
Regards
Juri
2003 Nov 18
0
Swissvoice ip10s MGCP questions and experiences
Hi there,
here some questions and experiences after playing for one day with 3
Swissvoice ip10s and the latest * CVS:
QUESTIONS:
- what is the user option "enter voice mail number" good for? It doesn't
appear to be of any practical use
- does anyone have some Swissvoice info that I cannot find on their web
site like the guide to MGCP XML (.svd), guide to configuration file
2004 Aug 23
0
Swissvoice MGCP Error 502
I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone.
When I dial the number for the IP phone off the POTS phone, the IP phone
rings. But when I pick up the
handset on the IP phone, I get a busy signal and this message on *:
Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response:
Terminating on result 502 from svip10@00059002042b-1
Here is the entire session. svip10 is the 1 and
2003 Oct 30
4
SwissVoice MGCP IP10S
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Any ideas are appreciated.
Robert
mgcp.conf is:
[general]
port = 2427
bindaddr = 192.168.0.110
[ip10]
host = 192.168.0.5
context = from-sip
line => aaln/1
The portion of extensions.conf is:
exten => 3001,1,Dial(MGCP/aaln1,20)
exten => 3001,103,Hangup
2006 Feb 15
2
Hint priority
Hi All
Has anyone managed to get the hint priority with Swissvoice IP10S phones
working?
I have 2 phones: a Snom 360, setup as the reception phone on extension
11, and a Swissvoice IP10S on extension 12.
When calling each other (tested both ways) I can only ever see the Snom
360 in the Active State from 'show hints'. The Swissvoice stubbornly
remains in the Idle State when on a call!
2005 May 15
1
Problem with extensions and when channel is unavailable
Hello
I used to have an extension like this which worked fine with asterisk
1.0.7
I first dial to see if an IAX phone is present, if not I would try on
SIP instead
exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout
exten=s,2,Goto(s-${DIALSTATUS},1)
; Default action
exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not
existing, goto 301
2004 Apr 06
6
swissvoice ip10s
hallo,
does anybody successfully managed to get swissvoice ip10s with h323
firmware work with asterisk ? mgcp firmware works fine, but with h323
i'm still getting one way audio.
regards
Marian
--
SUNTEQ s. r. o.
Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic
Tel: +421-46-5430 754 # Fax: +421-46-5439 144
http://www.sunteq.sk/
2004 Jul 13
2
Swiss IP10S using SIP
Has anyone had success getting the Swiss IP10S and the SIP ( IP10 SP
v0.0.1 (Build 4)) firmware working with Asterisk? If so do you have
copies of what worked in sip.conf and phone configuration files?
I can't seem to get the phone to register, it tries but is denied with
a Forbidden (which I am guessing is authentication). I tried without
a secret, but the phone seems to use swissvoice
2004 May 19
1
Swissvoice ip10: No 3-way-calling! (MGCP)
taken from bug 881 (now resolved) :-(
----------------------------------------------------------------------
markster - 05-19-2004 09:21 CDT
---------------------------------------------------------------------- As
it turns out the 10S cannot conference on the device. From Jean-Francois
at Swissvoice:
Hi Mark,
IP10S have not the capabilities to mix by itself 2 RTP flows, that why it
refuses
2004 May 04
1
MGCP: Current CVS works for you?
Hi there,
I have serious problems with MGCP and Swissvoice ip10s, and it appears
that recent CVS also introduced trouble for other MGCP users. Please
check and add comments in the bugtracker so that we can get a clearer
picture - thanks! Also comment if things are working fine for you.
http://bugs.digium.com/bug_view_page.php?bug_id=0001542
2003 Nov 19
1
Service codes for MGCP channels
Hi there,
after testing with a MGCP phone (Swissvoice ip10s) I found the following
ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that
most of those will also work with SIP, but haven't checked that yet:
*67 - Calling Number Delivery Blocking
*70 - Cancel Call Waiting
*72 - Call Forwarding Activation
*73 - Call Forwarding Deactivation
*78 - Do Not Disturb Activation
2007 Jan 19
1
Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to "yes" in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
2003 Nov 04
1
Call Transfert with SwissVoice IP10S in MGCP mode
Hello,
Now that I have a nearly working configuration for my IP10S with * I
wonder if anyone has done call transfert with this Phone. In the IP10S
documentation they talk about the 'service key' wich is the key with the
white dot on it. With this Key, it should be possible to have a menu
with call transfert entries. This menu should (accordingly to the
documentation) depend on the
2006 Feb 20
1
Dial timeouts and SIP 302 redirects
I have SIP handsets which allow the user to forward a call to another number
after a specified interval of ringing time. On the SwissVoice this is
refered to as CFNR (Call Forward on No Response). What actually happens is
that after a specified period of time (default 15 seconds), the handset
sends back a "302 Moved Temporarily" response to Asterisk.
The problem is that when Asterisk
2009 Jul 11
0
MACRO-INCOMING-CALL-TO-EXTENSION
Hello my friends,
I've a doubt, i want to be able to forward the incoming calls from PSTN to
my cell phone...i mean, qhen i'm out of the office i need like aq macro that
helps me to forward the incoming call that goes for example to my internal
extension SIP 207, i 've this macro but i can make it work properly....i
can't activate the forward in the phone, is quite confuse:
2005 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO. Incoming calls work fine, outbound I get this:
-- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack
-- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2007 Aug 28
1
calls being forwarded to neighbor?? please help, thx :)
hi ppl :D
my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2,
i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a
grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have
a machine (machine 1), which functions as my router and machine 2 and sip device are behind it,
grandstream box
2008 Jul 29
0
Fallback on a fallback
I have two sites running Asterisk PBX. Normally the inbound calls go
through a 3rd (colocated) server and are routed via IAX to the site
(the site registers with the main server)
I created a macro that tries to ring one location and then another.
Each site explicitly Answer() the call even though it will only ring
all the sip phones at the relevant location. When fall back is in
effect it goes to