similar to: Problem with demo on asterisk

Displaying 20 results from an estimated 5000 matches similar to: "Problem with demo on asterisk"

2003 Nov 21
9
Outline For Asterisk Book - Please Review & Comment
Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an "ebook" that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly & Co.). Below is the outline for the book. We REALLY
2003 Dec 15
2
Beginners Question
Hi all, New user to asterisk having just got it compiled and installed. Running with no digium hardware (yet) and no soundcard in asterisk box. Problem is using the sample configs with a sip phone added as follows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to
2005 Mar 03
2
FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls
2003 Nov 18
3
"Unable to find path from G729A to ULAW" on Sipphone.com
I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File
2006 Nov 13
1
Dial : Executing context/priority after bridge?
Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder. It obviously results in
2004 May 18
2
registering in sipphone
for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^
2009 Apr 26
1
1.6.1: "DNS error" but ping works
With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS
2006 Apr 26
1
getting asterisk to reliably answer a voip line
I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is
2005 Feb 01
2
Outbound calling with TDM400P
I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer
2003 Aug 06
4
New SIP Phone
Michael Robertson, founder of both MP3.com and Lindows, has launched a new company to supply inexpensive SIP phones ($129 for two) and related services. See today's press release at http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc My question for the list is who will be the first to report on the compatibility and usability of the SIPphone with Asterisk? The functionality
2010 Feb 17
1
One-Way Audio after Hold
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though
2004 Oct 05
2
SIPphone All-in-One: coments anyone?
Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help
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
2004 Feb 03
1
sipphone dialing out problem
Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing
2004 Apr 26
1
Problems registering with Sipphone
Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123
2006 Oct 20
2
modprobe Ztdummy is not working
Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point "modprobe ztdummy" it's failing. I'm getting these messages: root@dubic:~# modprobe ztdummy FATAL: Error inserting ztdummy
2005 Oct 13
2
ztdummy build problems
Hi all, Trying to build ztdummy on an old redhat 7.3 box running kernel 2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes, I set them up with make oldconfig; make dep. The build error is: make ztdummy gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/
2006 Mar 04
2
Problem compiling ztdummy on centos 4, 2.6 kernel
Hi all, I hope everyone is doing well. I just joined the list, and I've really enjoyed all I have read about asterisk so far. Unfortunately, I'm having a bit of trouble implementing this thing :). By the way ... I did my best to search the forums, and also to use google extensively, and while I have found pages with people with the same problem, ... The fix suggested on those sites,
2003 May 29
1
a beginner's SIP question ..
I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing