similar to: Legacy Cisco ATA186I1

Displaying 20 results from an estimated 9000 matches similar to: "Legacy Cisco ATA186I1"

2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G
2010 Mar 01
1
Fwd: Erika DeBenedictis-Recommendation
---------- Forwarded message ---------- From: Celia Einhorn <celia.einhorn at gmail.com> Date: Wed, Feb 17, 2010 at 8:15 PM Subject: Fwd: Erika DeBenedictis-Recommendation To: drew einhorn <drew.einhorn at gmail.com> ---------- Forwarded message ---------- From: David H. Kratzer <dhk at lanl.gov> Date: Tue, Feb 16, 2010 at 9:24 AM Subject: Fwd: Erika
2011 Jun 09
0
Insert name in SIP registry
Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600e\x0d\x0a From: Jian Gao <sip:8181234567 at my.provider.com>;tag=7e9c4091bfc704bco0\x0d\x0a To: Jian Gao <sip:8181234567 at my.provider.com>\x0d\x0a Call-ID: daf96244-769f952c at 10.11.22.161\x0d\x0a CSeq: 48998 REGISTER\x0d\x0a Max-Forwards: 70\x0d\x0a Contact: Jian Gao <sip:8181234567 at
2009 Jan 16
0
No subject
The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but
2009 Jan 16
0
No subject
The cell companies are "doing it" like they think makes sense. If they know your cell is off/out of range they route instantly to VM. They could give "4-10 rings" of fake effort, but why. With follow me roaming and such, they want to process the call as fast as possible. If they don't know if the cell is available, they may go through about 4 rings of searching, but
2006 Nov 20
2
Fwd: Traffic Shaping on a Transparent Bridge not working!
I''m trying to shape traffic on a Devil-Linux box. This note was originally sent to their maillist, because the LARTC list appears to have been down for the past few days. My mailbox was just flooded with a half dozen or so confirmation requests in response to my repeated attempts to subscribe to this list. ---------- Forwarded message ---------- From: drew einhorn
2009 Apr 09
3
T.38 ATAs
Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian
2010 Nov 21
0
How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk
I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out
2003 Jul 16
1
Cisco 7905G vs ATA186
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
2010 Feb 05
0
Do the Linksys Sipura series have a known problem with Asterisk?
I have a Linksys Sipura SPA2102 connected to Asterisk 1.4.27 and sometimes it doesn't connect at all. I keep getting a busy signal when I try to dial. It appears to happen most often when both lines are registered. The 2 lines on Linksys lines also use different ports. Does that mean than it is necessary to configure the 2nd line with port=5061 in sip.conf, or can they both use 5060? Thanks
2010 Mar 04
1
time/date over POTS?
I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j
2003 Sep 04
2
cisco ATA186 I2 vs I1
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:
2009 Mar 16
2
t38 iax trunk
Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys
2008 Jun 07
1
Software raid tutorial and hardware raid questions.
I remember seeing one with an example migrating from an old fashioned filesystem on a partition to a new filesystem on a mirrored lvm logical volume but one only one side of the mirror is set up at this time. First I need to copy stuff from what will become the second side of the mirror to filesystem on the first side or the mirror Then I will be ready to follow the rest of the tutorial and
2008 Nov 19
0
net-snmp puzzle
I have a bunch of centos 5.2 boxes. I'm trying to run smtp on all of them. Using indentical net-snmp configuration files. /etc/snmp/snmpd.conf com2sec notConfigUser localhost smssnmp com2sec notConfigUser 10.1.1.0/24 smssnmp group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view systemview included
2011 Apr 06
1
MWI not working on most ATAs in Asterisk 1.6.2.17
We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. While not exhaustive, these are the ATAs that don't work: Linksys SPA2102 Linksys PAP2T-3.1.15 Thomson 780 Thomson 784 Unfortunately, this
2010 Sep 06
1
Asterisk Fax
Hi I know that this topic was on the list maybe dozen of times. But I have a question regarding the fax support in asterisk, because all the information I could get does not give me the clear view of if. I read that Asterisk 1.8 will have strong fax (t.38) support, but I want to know if these four scenarios will be possible to achieve: fax machine (phone+fax) connected to ATA --- SPA2102 ATA ---
2004 Apr 15
1
Calls to Cisco PSTN gateway
Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp:
2004 Aug 05
1
is asterisk and/or spandsp what I need for integration with Cisco?
Hello list. I've been running a Cisco VOIP WAN for the last two years using CallManager V3.2. The remote 1760 routers are set up as H323 gateways. I've got a VG200 that is my PSTN gateway with a full T1 voice trunk to the local CO. I'm also running Unity v3.x for voicemail. We are currently using two ATA186 devices that are using the G.711 codec to pass incoming calls on the T1
2005 Jul 17
0
Cisco ATA186 Internal Dialplan: How to send *8?
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