similar to: Hang on "making progrogress passing" when dialing out

Displaying 20 results from an estimated 1100 matches similar to: "Hang on "making progrogress passing" when dialing out"

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 09
1
Dialing 800 numbers through FWD or SIPphone?
Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"
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 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
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}@??????) ^^^^^^
2005 Jan 27
1
CallerID for incoming SIP calls to Asterisk connected phone
Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The
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
2005 Feb 09
0
Asterisk and SIPphone won't cooperate
When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend
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
2009 Feb 17
0
Optimizing this script for calling Skype users from Asterisk
I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence is username at opensky.gizmo5.com but that gets mapped to sipphone address so I set that up to map
2006 Mar 15
2
Help with Gizmo from outside firewall
I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware
2005 Jan 03
0
Re: Asterisk won't register with sipphone.com
Hello All. I started setting up my Asterisk system yesterday and everything was going well, i have registered with sipphone.com and set-up my Asterisk system to register with sipphone per the sip.conf file below. It was registered perfectly but I could not receive calls so I added in the line "insecure-very" and I then used the Washington DC access number to test and the phone
2006 Feb 10
0
calling to sip provider
Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. "sip show status" (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's
2005 Jul 29
0
ReInvite X Broadvoice
I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and
2003 Nov 30
1
Dial "T" option not obeyed with Grandstream BT101
In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user
2009 Aug 05
0
Asterisk with gizmo5 and google voice only takes one call at a time.
my problem is this. I have google forward the call to gizmo5. I have this line in my sip file : register => user:password at proxy01.sipphone.com I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it? At the end of my sip file i have this [Calls-From-Gizmo-Network] type=user context=demo disallow=all allow=ulaw allow=ilbc allow=gsm
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
2009 Mar 19
0
Can I tell if a call picked up on PSTN extension... for example?
Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are
2007 Mar 25
1
Importing cds in OGG & FLAC in itunes/OSX
Hi all , Just to be sure : even if we can now use XiphQT to have import/export quicktime function on mac OSX for vorbis and theora , we still can't tell itunes to import a CD in Ogg vorbis , if I have understood ? is this because itunes handle certains things like playing streams and encoding Cds in a non quicktime standard way ? --