Displaying 20 results from an estimated 30000 matches similar to: "Issues with Asterisk & siproxd"
2005 Jan 24
1
Asterisk -> static nat -> laptop w/siproxd -> cisco 7960
Ok, I have a 7960 that's plugged into my laptop. my home network is
wireless so I don't have a switch anywhere to plug the phone into
directly. I'm running siproxd on my OS X laptop and I can make
outbound calls from the 7960 fine (I guess I don't have the phone
configured to register inbound calls via SIP), but the phone isn't
registering to the asterisk box via siproxd
2004 Jun 08
3
Sending # and Asterisk Transfer Conflict
Having spent the better part of an hour searching the archives and voip-info
I hesitantly ask what appears to be an obvious question but one I cannot
find an answer for.
Using Grandstream phones it seems that the only way to support Call Parking
is to enable # transfers (i.e. use T in the dial command) since pressing the
TRANSFER button on the BT phone is blind and one does not hear the call
2007 May 16
5
Microsoft's Move Into IP PBX Market
From c|net News:
"On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office Communicator 2007."
http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20
--
2007 Dec 31
1
PRI Crapping Out Regularly
We have a server with a TE120 on a partial PRI trunk that several times
a day declares the PRI trunk down and stops handling calls until the
asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk
started.
Just before things go down, the log shows the following error:
ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500
at which point a "show pri spans"
2005 May 09
2
AGI - How to Make Calls and Bridge to Original Incoming
I need to accept an incoming call, make a series of outgoing calls, and
once I find someone willing to accept the call, bridge the original
incoming call to the outgoing call.
Using Dial from an AGI script isn't enough because once the Dial'ed
number connects, the call is immediately bridged and I need to ask the
called party if they will accept the call.
I can see a couple of
2006 Mar 17
0
One-Way SIP Audio with SVN Codebase
Please tell me the obvious mistake I'm making here. (And yes, I well
know about NAT and one-way audio problems in general.)
I want to try the new T.38 passthrough stuff, downloaded it, built it,
tested it with an SPA-2100 and can hear announcements fine but echo test
shows no audio outbound (i.e. SPA to Asterisk).
Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in
2006 Oct 16
7
tdm2400p question
Hi all,
I'm confused, in digium website, it says:
TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a
total of 24 lines.
6 plus 6 is 12, how come it's 24?
if I have 24 PSTN lines, i'll be needing 24 FXOs.
Pls. elaborate.
thanks.
Lito
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2007 Feb 08
2
Suppliers in Canada
I am looking for some Linksys and GrandStream ATAs in Canada. I am
looking for places that ship from Canada so I don't have to deal
with the clearing of customs and tax remittance.
Any suggestion?
--
Thanks
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX
2003 Nov 21
0
One way sound
Hi,
I'm having trouble with asterisk: I can't hear both way of a call.
here is my current architecture:
grandstream -> siproxd -> asterisk -> pstn
As I'm just testing for the moment, evrything is on my LAN. I know that
there is no need to have a proxy here. But Later, the asterisk will be
on a public IP outside of my LAn, so I'm practising...
And there is no direct
2006 Mar 17
1
One-Way SIP Audio with SVN Codebase (CANCEL)
I wrote earlier:
> Please tell me the obvious mistake I'm making here....
The problem was a lack of sleep. Sorry to have troubled the list.
--
George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
www.netvoice.ca www.ip-centrex.ca
www.digium.ca www.grandstream.ca
2008 Jan 29
1
PRI Alarms, Comes Back, But Asterisk Won't Touch It!
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
(same problem with various previous versions; same problem with
different TE120P cards).
The customer has a partial (10 B-Channel) PRI that when it is busy
(eight or more B channels in use), tends to fail as shown below...
[Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown
error 500
[Jan 26 23:00:31]
2003 May 07
2
SIPPROXD for SIP thru NAT
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: siproxd.url
Type: application/octet-stream
Size: 82 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/bddd870b/siproxd.obj
2007 Feb 08
11
Best phone for easy provisioning
Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones,
and they're great (but too expensive). I like Grandstream phones, but
their provisioning sucks.
What is everybody else using in large environments where individual
config is not an option?
----------------------------------------
Rod Bacon
2007 Jun 08
0
Replacing SX-2000 Centigram Voicemail with Asterisk?
We have a customer with an obsolete Centigram voicemail system who would
like to replace it with Asterisk.
Any one with experience doing this or information on the signalling and
trunking used to connect the Mitel SX-2000 to the Centigram server?
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca
www.digium.ca
2005 Aug 24
0
Distorted Sound from E1
We're having a problem with an E1 trunk in Mexico into an IVR server and
would appreciate any suggestions.
Hardware: Digium TE110P jumpered for E1
zaptel.conf:
span=1,1,0,ccs,hdb3
# clear=1-30
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
Circuit status is fine: Status: Provisioned, Up, Active
Calls are accepted by Asterisk without any
2006 Jan 19
0
AudioCodes Unreliable DTMF Detection
We're trying to use some AudioCodes MP104 FXO units as gateways to
Asterisk but cannot get them to reliably detect DTMF. Some landline
calls get most digits but some are duplicated. Some cell phone calls get
0% DTMF recognition.
Anyone with experience with these units have any suggestions? ABP
Technical Support has been unable to diagnose the problem and is now
sending random guesses and
2006 Mar 15
0
T.38 Passthrough testing -- IAX problem
Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to
pass some calls to another using IAX and attempts to use the Dial
command results in multiple messages "Out of idle IAX2 threads for I/O,
pausing!".
Since this server needs to support IAX I'll have to back out this
version and find another idle server to use to play with the T.38 code.
g.
--
George
2006 Mar 18
0
T38 Passthrough testing -- unknown media type error
We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347):
- we are using a Sipura SPA-2100 as the T.38 user device
- we are using a Patton SmartNode 2400 as the T.38/PRI gateway
- we are using Asterisk in the middle
We have the following in the [general] section of our sip.conf:
t38pt_udptl = yes
t38pt_rtp = yes
When a fax call comes in from the SmartNode to Asterisk
2007 Jul 12
0
No subject
"Annoying that people aren't following the directions and only entering 3
digits, but we've had some high level meetings here with a string of clients
coming through in an unusually compressed frequency. And I've had 5
complaints over 2 days that callers couldn't find Jane Smith."
-
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102