similar to: sip ata's

Displaying 20 results from an estimated 9000 matches similar to: "sip ata'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 Jun 11
0
Problems configuring Asterisk with SIP
Hi everybody Could someone give a tip on how can I configure asterisk to use 2 ATA's 186 to communicate each other using SIP with asterisk. I know this most be a very simple task, however this is the very first aproach I have to asterisk. I set the following config but I don't get dial-tone when I off-hook the phone from any of the two ATAs. Can some one tell what I'm missing in
2006 Dec 15
1
Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?
Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured
2006 Mar 30
0
Strange second REINVITE being sent
I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid="ATA 1234567" host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When one ATA calls another, I see the next traffic on Ethereal (just shown the sequence
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
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
2003 May 17
0
Debug for SIP and reINVITES (ATA-186)
I must be doing something incorrectly, or something is wrong with ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me can lend me a hand. I have been attempting to get two SIP phones to reINVITE to each other, and I've been unable to think of or uncover the correct method. The calls always go through the Asterisk server, no matter what I try. I've simplified things
2005 Jan 03
0
Echo problem - (sorry if this is an nmf question)
I recently installed * on my firewall and that of a relative some miles away. I route sipphone (kphone and x-lite) calls from deep within the backbone (two layers of firewall) on each end to the other. Works fine between @200Mhz pentium doorstop linux boxes (even w/2.4 kernel).The problem of course is the output of the speaker at the other end is picked up by the microphone (confirmed by
2009 May 28
1
asterisk 1.4.X, T.38 and NAT
Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same
2005 Jan 18
0
DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a
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
2006 Apr 10
0
Problem with Asterisk and Grandstream HT286
I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes
2003 Sep 08
1
extension.conf and SIP phones.
We would like to setup in house SIP phones with numbered extensions for demonstration purposes. What is the syntax to associate a extension with SIP phone? Does the Dial application have a SIP specific entry for example: Dial,SIP/SIPphone/s|15 When I call from one extension to another I get "User is on the phone". We also have Cisco7960s to test. Currently Have X-Lite setup.
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
2014 Dec 15
1
T.38 not working - help needed with log interpretation
On Mon, Dec 15, 2014 at 3:34 AM, Recursive <lists at binarus.de> wrote: > <snip> >> For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', I don't know what Asterisk 13 will do with this setting. >> > I suspect Asterisk 13 will just ignore it. To make things worse, there seems to be a configuration directive named reinvite (not a typo); I
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
2005 Sep 30
1
Music on hold not initiating RTP stream?
I've been having problems getting MusicOnHold to work, so I've dumbed down my setup to as simple of a setup as I can. Asterisk 1.0.9. SIP ATA's (Sipura SPA-2002's) <SIP ATA 1> <---> <Asterisk> <---> <SIP ATA 2> Both ATA's have public IP's. No NAT'ing going on here. Reinvites are allowed so the media stream bypases Asterisk once a call
2004 Oct 03
3
ATA's
Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm
2012 Aug 18
1
asterisk tries reinvite when incompatible codecs on call legs
Hi, I just ran into what seems to be an issue on re-invites. I'm not sure if it's a bug or as designed, so I thought I'd ask the question. Here's my setup: - Asterisk 1.8.13.0 - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes Phone A calls the extension of phone B. After the normal call setup
2019 Aug 16
2
PJSIP reInvite
Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood