On Wednesday 07 December 2005 11:10, John Voss wrote:
> I can't seem to get my outgoing connections to work with IConnecthere. At
> one time it did with v1.0
>
> I can register and receive calls just fine. But can't make them.
>
> Ultimately, the trace ends with a "400 Bad Request" error when you d
I can't seem to get my outgoing connections to work with IConnecthere. At one
time it did with v1.0
I can register and receive calls just fine. But can't make them.
Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP
debug.
Has anyone got it to work with v1.2? Don't kno
I've stumbled upon a very interesting phenomenon.
In setting up a trunk from iConnectHere (ich) I mistakenly input
type=from-pstn
type=friend
This, in [EMAIL PROTECTED] using AMP.
No "User Context" entries are made
and a DID entry (with the full 11 digit number) is in the DID settings.
I now
Once upon a time Wednesday 20 July 2005 10:42 pm, Rich Adamson wrote:
>
> Kind of sounds like the real question is "who is suppose to provide
> ringback when you are using asterisk"?
>
> My guess is that you are. Try "show application dial" and look for
> the "r" parameter, etc.
I guess so. Works
> I have service with iconnecthere they use sip and allow generic devices.
> outbound and inbound calls work as expected. i had a few hicups but nothing
> i couldnt workout with sip debugging turned on.
>
> I have one issue and its a doozie. I dont think its related to asterisk
> thoug
Hi All,
I have service with iconnecthere they use sip and allow generic devices.
outbound and inbound calls work as expected. i had a few hicups but nothing
i couldnt workout with sip debugging turned on.
I have one issue and its a doozie. I dont think its related to asterisk
though th
I try to use asterisk to make automatic autobound call
to do survey. this requires DTMF for feedback from
people called.
My setup is:
asterisk--->iconnecthere>PSTN
but asterisk is unable to detect any key pressed. No
information shows up in the CLI. I tried to set
dtmfmode=RFC2833, info and
Did iConnectHere ever fix their inbound DTMF problem? Is it useable with *
again?
- Original Message -
From: "ERwin Hernandez" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 05, 2004 12:41 PM
Subject: [Asterisk-Users] iConnectHere and CallerId
Is it possible to send the CallerId to IconnectHere with Asterisk
when making outbound calls?
I read somewhere that it doesn't work.
I have set up everything to send the correct CallerId info to IconnectHere
but I get a "442-887-926267" caller id.
In [globals]
ICONNECT1=1713...(my number)
MYNAME=
> I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped
> working. I
made no changes to my conf files or anything
> else (other than the upgrade). Now I get a "Unsupported Media" error even though I'm
> still
using ulaw and alaw.
>
> It stopped working with my sof
Title: iConnectHere broken?
I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a "Unsupported Media" error even though I'm still using ulaw and alaw.
It stopped
[EMAIL PROTECTED] wrote:
Check 0001436 in the bugtracker. This was the original bug fix which
broke outbound calls. Additional work was done on this bug to fix a
problem with incoming calls (see marks comments at the end). Maybe you
got a CVS while this was being worked on? Maybe there is still
Jr
Brian Capouch <[EMAIL PROTECTED]>
Brian Capouch <[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
05/03/2004 11:47 AM
Please respond to
[EMAIL PROTECTED]
To
[EMAIL PROTECTED]
cc
Subject
Re: [Asterisk-Users] iconnecthere behind NAT, strange deal
Lists w
Lists wrote:
Have you found a solution yet? I am having the same issue. My account works
fine with the IConnectHere soft phone client but not with Asterisk. Inbound
lines work fine, outbound returns the same message.
CVS update took care of my problem wrt outbound calls; it must have been
the mis-
Thanks for the update!
Michael
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, May 03, 2004 1:18 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
iconnecthere behind NAT, strange deal
Hi,
I had the same issue with
resolve the problem you are seeing.
Bill Doll Jr
">"Lists" <[EMAIL PROTECTED]>
"Lists" <[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
05/03/2004 08:09 AM
Please respond to
[EMAIL PROTECTED]
To
<[EMAIL PROTECTED]>
cc
Subjec
Behalf Of Brian Capouch
Sent: Sunday, May 02, 2004 4:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iconnecthere behind NAT, strange deal
I've been to the WIKI and I've searched the archives.
Is anyone on the list successfully using iconnecthere behind NAT?
I was, for over a year,
> Is anyone on the list successfully using iconnecthere
> behind NAT?
Yes (unless it broke during the last 12 hours).
I gave my daughter a SIP phone and an Iconnecthere account.
I have successfully used Sipura 2000, a Grandstream and a
Cisco ATA box with that account, and yes, she is behind a
*sta
I've been to the WIKI and I've searched the archives.
Is anyone on the list successfully using iconnecthere behind NAT?
I was, for over a year, and then I changed my "plan" with them. Now all
my calls get intercepted immediately, "We're sorry, but your account is
temporarily unavailable."
Inc
Hi guys
I just registered an incoming number with iconnecthere and I'm trying to
set up incoming calls from icconnecthere on my asterisk server. I took a
look at john todds sample sip.conf and extensions.conf file but for some
reason my incoming is still not working. At this point I wish to us
Can anyone provide me with a current config for recieving calls with
Iconnecthere? I'm having some difficulty with it...
Regards,
Phillip
--
Phil Jackson, President & CEO
The Jackson Group - Intelligent IT. (TM)
www.jacksongrp.com
___
Asterisk-Users m
Has anyone made * to work with iconnnecthere's demo account?
Todd Wallace
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ED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 27, 2003 8:49 AM
Subject: Re: [Asterisk-Users] Iconnecthere connect problem
> Hello..
> Thanks for the reply.. I'll give this a check later today. Is the first
> x in the register command your phone number at ICONNECTHERE?
tensions.conf:
>
> exten => _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
>
> This works for me
>
> regards
>
> Miklos
>
>
>
> - Original Message -
> From: "rnc Info Lists" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent:
egards
Miklos
- Original Message -
From: "rnc Info Lists" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, October 25, 2003 5:17 PM
Subject: [Asterisk-Users] Iconnecthere connect problem
> I have an Asterisk box behind NAT and am trying to connect to
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial("SIP/2001-12c8", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
== No one is available to an
When I make a call using iconnecthere, I hear no ringback tone, but the
call does get connected. Any suggestions?
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When I make a call using iconnecthere, I get no ringback tone, but after
the ringing the call does get connected. Any suggestions?
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Try
host=sipauth.deltathree.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Wednesday, September 17, 2003 6:46 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Iconnecth
bject: [Asterisk-Users] Iconnecthere Problem
I can't seem to dial out with Iconnecthere. I am using the following
commands.
I get a message that the session is in progress and the call never goes
through. Can anyone confirm if iconnect is working and if I am missing
something?
Thanks,
Ke
I can't seem to dial out with Iconnecthere. I am using the following
commands.
I get a message that the session is in progress and the call never goes
through. Can anyone confirm if iconnect is working and if I am missing
something?
Thanks,
Kevin
>
> in sip.conf:
>
> [iconnect]
> type=friend
>
On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote:
> On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
> > Does anyone have Asterisk working with Iconnect here for incoming
> and/or
> > outgoing calls?
>
> have a look at:
>
> http://www.loligo.com/asterisk/example-configs.2003-04-24/extensi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 5:29 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
> -Origi
Hi,
On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
> Does anyone have Asterisk working with Iconnect here for incoming and/or
> outgoing calls?
have a look at:
http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.conf
there's a section in there dealing with Iconnect
wkr,
Hi,
> I seem to have my configuration working except for outgoing and incoming
> calls for the rest of the world. For now I am concerned more about
> outgoing calls than anything else. Whenever I try to make an outgoing
> call I get these messages from the sip debug in the console
>
> s=session
>
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
> Verstappen
> Sent: Sunday, August 10, 2003 4:57 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asteris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 4:57 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote:
> On Sun, 2003-08-10 at
appen
Sent: Sunday, August 10, 2003 3:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Iconnecthere
Hi,
On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
> Does anyone have Asterisk working with Iconnect here for incoming
and/or
> outgoing calls?
have a look at:
http://www.lol
Does anyone have Asterisk working with Iconnect here for incoming and/or
outgoing calls? If you would be so kind as to share with me the
configuration you have used, as I cannot seem to get my SIP service to
work although it does seem to be registered with the other end:
hm6*CLI> sip show registry
Brad,
Great. I suspect my difficulty may be related to a NAT/PAT
configuration. The SIP/SDP negotiations go fine, but there could be
problems setting up the incoming RTP session. I wonder how SIP/SDP
decides on what port(s) to listen for the incoming RTP connection.
Gregg
On Sun, 2003-03-30 at 1
I'm not behind a NAT, but of course behind a firewall (duh). I was even
thinking to myself "this is very much like what happens with IAX when
there is a firewall issue". So having taken care of that, it works great
with the same sip.conf settings you have below, and both directions can
hear each ot
brad,
Just to make sure you understand the settings, not using the prefix
tells iconnect to use uncompressed codecs. Using sets iconnect into
compressed codec mode.
I am experience that same problem as you when I try to use the
uncompressed mode. I connect, but cannot hear the other par
I've tried these settings and I still find that I cannot hear the called
party. I've also tried what feels like every allow/disallow combination
with and without a prefix and I either get 488 errors, using one
format when the capability is another errors, or completed calls where I
can't h
>Is there a Record-Route header in the response that comes back from
>iconnect?
In the 480, not that I can see.
In the 408, I'm not sure as I didn't have SIP debugging enabled (and I don't
have anyone internationally to ring right now :-)).
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.
Mark,
I believe there is: Here is the exchange using sip debug.
Gregg
---
bigcat*CLI> sip debug
SIP Debugging Enabled
-- Executing Dial("Phone/phone0",
"SIP/[EMAIL PROTECTED]") in new stack
Interface is eth0
IP Address is 192.168.4.3
We're at 192.168.
Is there a Record-Route header in the response that comes back from
iconnect?
Mark
On Sun, 23 Mar 2003, Luke Howard wrote:
>
> >> Or maybe we should send an ACK to them -- I need to read the SIP RFC...
> >>
> >
> >Tried that, doesn't work.
> >
> >I should add that in my config I'm totally behind
Gregg,
>1) the prefix is not a toggle. It tells iconnects SIP gateway to
>use compressed codecs. The choices are gsm, g723.1, g729.
I figured as much. I'm sticking with G.711 as GSM sounds horrible (at least
with the snom phones) and the other codecs you mention are patent
encumbered.
>I
Luke,
here's some information I got back from iconnect:
1) the prefix is not a toggle. It tells iconnects SIP gateway to
use compressed codecs. The choices are gsm, g723.1, g729.
If you don't use , the gateway will tried to use PCMu/8000 (ulaw?)
or PCMa/8000 (alaw?).
I can get the gate
FWIW here's the patch I'm using to ignore 480s:
[EMAIL PROTECTED]/monk[16]% cvs diff -u channels/chan_sip.c
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.5
diff -u -r1.5 ch
>> Or maybe we should send an ACK to them -- I need to read the SIP RFC...
>>
>
>Tried that, doesn't work.
>
>I should add that in my config I'm totally behind NAT, both asterisk and
>an ATA186 that talks to it.
>
>So that may be confounding me in terms of what I'm seeing.
I do see the same pro
>I should add that in my config I'm totally behind NAT, both asterisk and
>an ATA186 that talks to it.
Hmm, both our SIP phones and Asterisk are on visible IPs.
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.padl.com
___
Asterisk-Users mailing
Luke Howard wrote:>>>I tried the grotesque hack of making
handle_response() ignore 480
errors, which *seems* to work. Hmm.
I tried that, and at least for me it has a number of subtle side effects:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
Tried that, doesn't work.
Moreover, if anyone has a packet trace of iConnectHere's SIP client
making a call (which presumably does work), then please send it
along... it would be interesting to see whether Asterisk, at fault
or not, can be made to work around this properly.
-- Luke
--
Luke Howard | PADL Software Pty Ltd
>> I tried the grotesque hack of making handle_response() ignore 480
>> errors, which *seems* to work. Hmm.
>>
>
>I tried that, and at least for me it has a number of subtle side effects:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
-- Luke
--
Luke Howard | PADL Soft
>> I tried the grotesque hack of making handle_response() ignore 480
>> errors, which *seems* to work. Hmm.
>>
>
>I tried that, and at least for me it has a number of subtle side effects:
>
>1. Calls all cut off after just a few minutes
Well, we could enable the hack only on outgoing calls (whic
Luke Howard wrote:>>I remember at some point getting 488 media errors if
I didn't enable
gsm.
As I mentioned, I'm getting 480 Temporarily not available, not
488 media errors.
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
I tried that, and a
>I remember at some point getting 488 media errors if I didn't enable
>gsm.
As I mentioned, I'm getting 480 Temporarily not available, not
488 media errors.
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
-- Luke
--
Luke Howard | PADL Softw
>GSM works but the voice quality is absolutely terrible. This is the
>case with or without the prefix. (Did anyone ever figure out
>whether is a toggle?)
One thing I didn't realise until reading the new documentation is that
the codec list is in order of preference. So, if there's an a
>I remember at some point getting 488 media errors if I didn't enable
>gsm.
GSM works but the voice quality is absolutely terrible. This is the
case with or without the prefix. (Did anyone ever figure out
whether is a toggle?)
>disallow=g723.1
>allow=gsm
>allow=ulaw
>allow=alaw
>allow
I remember at some point getting 488 media errors if I didn't enable
gsm.
Here are my sip.conf and extensions.conf entries. They work for calls
out to iconnect:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context=iconnect ;
Luke,
Try putting the prefix of before your phone number. It changes the
codec expected by iconnect.
Gregg
On Thu, 2003-03-20 at 18:25, Luke Howard wrote:
> I've found the same.
>
> If I make an outgoing call (snom 200 handset), I get about 5 seconds
> of audio and then it drops out (very
Just an FYI but I'm seeing the same thing using ata-186's
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
http://www.netrom.com
I've found the same.
If I make an outgoing call (snom 200 handset), I get about 5 seconds
of audio and then it drops out (very occasionally it does work).
Incoming calls appear to work, though.
-- Executing Goto("SIP/515-Office-143b", "iconnecthere-ulaw|91800XXX|1") in new
stack
-- Got
rt of resembles a double press. I have heard this with the different
dtmfmodes.
John
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 12, 2003 6:51 PM
Subject: Re: [Asterisk-Users] iconnecthere DTMF solution?
Probably you should do dtmfmode=inband in the general section.
Mark
On 12 Mar 2003, Matthew Farley wrote:
> Finally, I have NATted ATA-186s working with Asterisk (thanks to
> all who made this happen)! My final troubles were with the firmware
> version in the 186 -- if you have troubles wit
Finally, I have NATted ATA-186s working with Asterisk (thanks to
all who made this happen)! My final troubles were with the firmware
version in the 186 -- if you have troubles with this (as I did), make
sure you have the newest firmware in the 186.. Otherwise it just won't
work.
Now for
, March 04, 2003 1:09 AM
Subject: Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
> I get these errors (480 "Temporarily...") when I try to use my
> iconnect account quickly after hanging up on a previous session.
> They have some sort of contention locking
I get these errors (480 "Temporarily...") when I try to use my
iconnect account quickly after hanging up on a previous session.
They have some sort of contention locking system which allows only
one call at a time on an account, and if you do not give it adequate
time to "settle", you'll hit t
I am going to have to find a fix for this problem or I'm going to have
to quit using iconnect.
About one call in 10 or so, iconnect's gateway gives me an error
(console output appended below).
So upon receiving the error, which as a 4XX error means, "Fatal,"
asterisk gives up and drops the cal
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