similar to: One way audio after IVR tree

Displaying 20 results from an estimated 10000 matches similar to: "One way audio after IVR tree"

2009 Jul 09
1
Weird audio problem with remote IVRs + DMTF
Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some
2008 Dec 05
2
All lines occupied notification from endpoint
Hi, I've noticed that if I have a multi-line linksys (942 or 962) phone with the same sip registration mapped to each line key, that if all the lines are full the phone will accept another call. I would expect the phone to respond with "busy" so the call would to directly to voicemail. Has anyone else experienced this and know of a workaround? I know it seems like an
2009 Jan 27
2
Muted sound on a Linksys 962
Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to
2004 Apr 26
3
dtmf tone clamping in calls to external ivr
Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between
2009 Nov 24
2
audio cuts out during IVR
Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I
2007 Nov 14
0
IVR Tree Best Practices
Does anyone have any solid documented evidence for best practices for IVR Trees? I'm just curious. I did some google searching this morning, but only found one article from TMC. What I'm looking for are studies showing, if a customer must go to an IVR, what they prefer, over what they prefer to not have in the trees. -------------- next part -------------- An HTML attachment was
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
2006 Jun 16
0
no IVR audio but phone to phone fine
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I'm having trouble getting my IVR to produce audio onto my cisco 7940 handset using SIP. I've got CentOS 4.3 running * 1.2.9.1 and the corresponding latest releases of libpri, zaptel, addons and sounds. I'm using two extensions to test that successfully dial each other and can talk back and forth fine. When I dial into the IVR, I get a
2008 Mar 25
0
Distorted Audio for incoming DTMF
Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture
2006 Nov 09
1
DTMF problems with IVR - What DMTF Tx method
I'm having problems with a new asterisk PBX install. the phones/ATAs are all linksys/cisco. They all worked before with a commercial softswitch. Most of the linksys devices offer auto, inband, INFO and AVT. I'm looking for suggestions. Thanks in advance -- One day at a time, one second if that's what it takes
2006 Jan 12
2
DTMF Issues With Asterisk 1.2 IVR
Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. --------------
2011 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>
2005 Mar 04
0
ANNOUNCEMENT: Updates for app_cbmysql andMeetMe2gui (out of tree modules)
First off, let me thank Bela?d Arezqui (aka Areski) for his PHP gui. I knew nothing about PHP last week, and the code makes for easy editing and additions. >Lots of interest here for conferencing. >I've probably convinced more people to start using asterisk@home for >this feature than anything else. >Can I input some suggestions; Sure >Need to change the rinky dink call
2012 Feb 25
0
No IVR audio. Jump in RTP sequence number
My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a
2005 Oct 16
1
No Audio from Console but mpg123 from shellworksfine.
-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. >One possibility is that the volume is set to 0. aumix can be handy here. Does
2008 Jul 03
1
CentOS 5.2 vs. Intel 82801 AC'97 Audio
I have installed 5.2 on my work desktop and everything works perfectly EXCEPT the sound device. Here is the relevant lspci line: 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) I tried looking in Google (might have been a bad search), but all I could find was to add acpi=on to my boot line in grub.conf, and that
2009 Jun 26
4
T38 Fax Gateway for Asterisk 1.6
Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James
2008 Jul 07
1
Flaky desktop audio behavior.
Might have some kind of bug here. I want to know if anybody else is seeing this. It's happened twice now, so I should be able to repeat it. $ rpm -q centos-release centos-release-5-2.el5.centos On my desktop, I "su -" to another user. Then "nohup thunderbird", it gets the mail and plays the announcement. ISTM this is an error right off the bat as this user is not the
2008 Mar 19
1
Getting config from SPA-941 or 942 phones
Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.
2014 Nov 25
2
High resident memory with 11.14.0 ?
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >