similar to: strange caller id and caller name with SIP and ATA186

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

I've got my ATA's setup with DHCP, host=3Ddynamic, and turned the registration on in the ATA All is well as long as my sip.conf is configured using the phone number like so... [1231231234] type=friend host=dynamic context=main canreinvite=no username=1231231234 secret=worknow disallow=all allow=ulaw allow=alaw It appears to me that the Cisco ATA (v3.1) will register the device using the

I have a Vonage acct that uses the Cisco ATA186. Currently, I have the ATA186 plugged into a SPA3000 to act as the FXO port. I installed a X100P card with the idea of replacing the SPA3000. Now, when I plug in the ATA186 into the X100P card and make a call into the system (from cell phone) and hangup when the IVR is playing, Asterisk is not detecting a hangup and keeps looping the IVR. If

Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 --> asterisk --> ATA186 --> FXS to FXO Converter --> PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks

Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) Thanks Duane Cox -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

Hi, I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance. I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1 version. Will the I2 version work in Canada with regular anlog phones, or will I need to change it. Thanks for your answer. Samy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is

I have done my homework on this, I hope. I have a customer with an ATA186 who uses Nufone as his IAX provider. His network operations center in the Bahamas was destroyed by the hurricanes, and I'm helping him rebuild. We have a nagging problem getting his ATAs (located in public IP space) to talk through his IAX provider (Nufone) to the outside world. As far as we know, things worked OK

This deal has got me confused. My dial plan rings my ATA186 on all incoming calls. If I don't pick up, it goes to voicemail. Under either of those circumstances, the callerid screen on my phone stays blank, and the message waiting indicator does NOT come on. But anytime a call comes in for me while I'm already talking on the phone, BOTH of those things happen. . . So what do I

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

All, I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work fine but outbound faxs receive congestion from *. I've got packet dumps from both sides and everything appears normal but after about 3 seconds the * servers sends the AS5300 a CANCEL and sends the ATA a '503 Service

Hi, Using latest cvs. I (as many otheres it seems) can't get Attended transfer to work with Cisco ATA186 (using SIP) Has anyone else had any luck? Same with 3-part calling, if one drops off, all are disconnected... /Stig

I have been beating my head against the wall trying to get my ATA186 to send through the *8 (call pickup) sequence back to Asterisk. The Administrator's Guide from Cisco would indicate that the first element in the default dialplan *St4- would mean that any sequence of digits following a * would be sent out after a 4-second timeout. But if I hit *8 it just sits there forever. If I hit