[Asterisk-Users] Cisco ATA-186 working peer to peer
Hi List. Can anyone can tell me if I can connect 2 Cisco ATA-186 in a peer to peer layout (without an Asterisk server registerisng the devices) through Internet? If it´s possible, could you help me with the configuration ? Many thanks in advance. Luis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 working peer to peer
Hi Luis, Can anyone can tell me if I can connect 2 Cisco ATA-186 in a peer to peer layout (without an Asterisk server registerisng the devices) through Internet? If running MGCP or SCCP, no. If running H.323 or SIP, and both ATAs are on static public IPs, no problem. Just specify the address of each unit as the gateway or proxy for the other. Disable registration. If NAT and/or dynamic IP is involved, it depends on what firmware version you are running, whether the NATs are aware of the protocol being used, and whether you have administrative control of them. But, why are you trying to do this? If you just register the two units with Free World Dialup or similar, it should work ok with NAT and dynamic IP, and the config will be provided for you. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 with *70
Has anyone come across the Cisco ata 186 not passing *70? WhenI press *70, the ata just goes back to a dial tone. The strange thing is, its only *70 and not the reset of the 70's. *71, *72, etcall go through fine. I've tried removing thethe dialplan all together from the ata to try and let it pass, but no go??? If I setup an IP phone to its extension, *70 goes through fine, so i know its definatly the ata. Anyone know away around this? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186 with *70
I seem to be having the exact same issue with the cisco ata 188. Not sure, but looking at the cisco manuals there are alot of options in hex format one can add, though 0x should cover all. On Mon, 9 May 2005, Christopher Kenna wrote: Has anyone come across the Cisco ata 186 not passing *70? When I press *70, the ata just goes back to a dial tone. The strange thing is, its only *70 and not the reset of the 70's. *71, *72, etc all go through fine. I've tried removing the the dialplan all together from the ata to try and let it pass, but no go??? If I setup an IP phone to its extension, *70 goes through fine, so i know its definatly the ata. Anyone know a way around this? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 and Asterisk
Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186 and Asterisk
I think it has to do with your CallFeatures. Callfeatures: 0x I have a screen shot of my converters config if you want it, it supports call waiting. I had to turn it off on one of my customers converters once, I had to change the last 2 digits or something to turn off call waiting. But it's on by default. What I found interesting is that ATA-186's were originally designed by Sipura for Cisco, according to something I read. I was wondering why linksys's latest converters were made by Sipura, seeing as how their parent company Cisco already made SIP converters, but it makes sense now, Cisco just went back to Sipura. --- Christopher Iarocci [EMAIL PROTECTED] wrote: Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186 and Asterisk
if you could set me up with your config, that would be great. thanx Chris [EMAIL PROTECTED] 5/7/2005 6:52 PM I think it has to do with your CallFeatures.Callfeatures: 0xI have a screen shot of my converters config if you want it, it supportscall waiting. I had to turn it off on one of my customers converters once,I had to change the last 2 digits or something to turn off call waiting.But it's on by default.What I found interesting is that ATA-186's were originally designed bySipura for Cisco, according to something I read. I was wondering whylinksys's latest converters were made by Sipura, seeing as how theirparent company Cisco already made SIP converters, but it makes sense now,Cisco just went back to Sipura. --- Christopher Iarocci [EMAIL PROTECTED] wrote: Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186
I'm nothing understand now. I have Cisco ATA 186 with one analog phone and
the following problem:
The next config works just fine:
sip.conf:
[150]
type=friend
port=5060
context=officepbx-outgoing
qualify=yes
secret=password
user=150
username=150
fromuser=150
defaultip=XXX.XXX.XXX.XXX
host=dynamic
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=alaw
extensions.conf:
TEST = SIP/[EMAIL PROTECTED]
agent_150 = ${TEST}
exten = 150,1,Macro(callfullext,${TEST},,,30,N)
But if I rename 150 to Cisco, by example, I get the following error
message:
NOTICE[1174440880]: chan_sip.c:7519 handle_request: Registration from
'sip:[EMAIL PROTECTED];user=phone' failed for 'XXX.XXX.XXX.XXX'
sip.conf:
[Cisco]
...
extensions.conf:
TEST = SIP/[EMAIL PROTECTED]
...
Cisco configuration:
UID0: 150
PWD0: password
UID1: 0
UseLoginID: 0
What is going on?
--
Serge Matveev
Relcom Corp., St.Petersburg
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[Asterisk-Users] Cisco ATA 186: Only incoming - no outgoing call
Hi all, I have configuredmy Cisco ATA 186as explained in John Todd's Guide: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt i.e, I have the following in my sip.conf [2299]type=friendusername=2299secret=blahblahcanreinvite=nohost=dynamicdtmfmode=rfc2833 And I have in my extensions.conf exten=5,1,Dial(SIP/2299,20,tT) I am able to call the Analog phone (Alcatel) connected to the ATA, from my X-Lite softphone, but not able to call in the reverse i.e., from the Analog phone to X-Lite! I am not seeing any error messages on Asterisk console. What can be the problem? Do you Yahoo!? Better first dates. More second dates. Yahoo! Personals ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 for PSTN dialing
Hi all.. can I configure Cisco ATA 186 to dial out to PSTN? I need a quick and easy to set up scenario to have SIP phones dial PSTN numbers. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA-186 and Caller ID
I'm having an interesting issue with the caller id generation of the Cisco ata-186. When the information is displayed, the name is displayed properly yet the number is corrupted, I get several solid boxes followed by a one. The ATA is set to the default bellcore as it recommends for use in the United States. Any suggestions on what to look into? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 and Caller ID
On Sun, 17 Oct 2004 19:52:50 -0400, Cory Andrews [EMAIL PROTECTED] wrote: Michael Greb wrote: I'm having an interesting issue with the caller id generation of the Cisco ata-186. When the information is displayed, the name is displayed properly yet the number is corrupted, I get several solid boxes followed by a one. The ATA is set to the default bellcore as it recommends for use in the United States. Any suggestions on what to look into? Michael - Which version of the Cisco ATA do you have, is is the I1 or I2 version it should say on the back of the unit. l1, running firmware v3.1.0 atasip (Build 040211A) Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- There is a configurable option within the 1204 to disable silence suppresion or something like that. As I recall, the option is configurable on a per-port basis. That option has to be disabled. (As stated earlier, I no longer have the 1204 so can't look up the actual parameter.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA 186
I had the same problem with a Mediatrix, it turned out to be a defective unit. No matter what we did the audio was very choppy, when I replaced the unit my problems went away. Are you running it as SIP or MGCP? Norm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gonzalo Gasca Sent: Wednesday, July 21, 2004 12:22 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA 186 Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186
Gonzalo Gasca wrote: Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? Turn VAD off on the 1204. * can not clock itself. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 from iconnecthere, locked?
I wanted to sign up for the pay as you go plan from iconnect anyway, and see they have the Cisco ATA for $99 and the Grandstream phone for $39.00 Anyone know if they ship these devices locked? I know iconnect seems pretty friendly about letting any sip device connect. What sucks is there is no way to contact this company if you're not a subscriber.. Zip, notta.. No email address, phone number, nothing. I guess I could send them a fax.. :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ata-186 port died
This might be the problem. I remember that i turned of the ringer (its an older style telephone with a switch on the back to switch to pulse, and turn the ringer off) so maybe the ATA had to much resistance and blew something. Anyone have expirience with this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ata-186 port died
I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was unsucessful in finding anything on google and wiki about it. jacob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA-186 Firmware upgrade
I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA-186 Firmware upgrade
There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 Firmware upgrade
im interested if there are any codec adds or major things like that... On Jun 12, 2004, at 6:44 AM, usedcanon wrote: There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco ata-186 behind NAT
i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions, i've been able to get the device to register to asterisk, and make calls without any problem. however, the asterisk box cannot see the adapter, and does not respond to hangup requests (therefore it would seem that the rtp stream is working properly in both directions, but SIP traffic is not finding it's way back). i have been focusing on two parameters in an attempt to get things functioning normally - namely NatTimer and ConnectMode. I have the following settings currently: ConnectMode: 0x20460400 (have also tried what i've seen elsewhere - 0x00460400, and 0x01a40400) NatTimer: 0x0054000a I've also tried the defaults and anything else suggested by others. If anyone has an ATA-186 running in a similar configuration and could share their configs with me it would be greatly appreciated. Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco ata-186 behind NAT
On Wed, 2004-06-02 at 15:40, Steven Kokinos wrote: i have been focusing on two parameters in an attempt to get things functioning normally - namely NatTimer and ConnectMode. I have the following settings currently: ConnectMode: 0x20460400 (have also tried what i've seen elsewhere - 0x00460400, and 0x01a40400) NatTimer: 0x0054000a This is the standard config we use for ATA-186s using v2.16 firmware: http://www.fnords.org/~eric/asterisk/ata-186.shtml We are slowly migrating to the 3.1 firmware, but the settings are very similar. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco ata-186 behind NAT
Try moving the ATA-186 to a port other then 5060. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 cannot make a call
Hello, We have a * box up and running with a handful of SNOM 200 phones. It is working very nicely. I am trying to add an analog phone via a Cisco ATA-186 box. The ATA-186 registers fine. It will receive a call. But when it comes to dialing out we get nowhere. :( The ATA-186 is @ firmware level 2.16.2. Are version of * is CVS-022504 and our Linux kernel is 2.4.20 (SUSE 8.2) Our end goal is to attach a fax machines to the ATA-186 Thanks Lach ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA 186 cannot make a call
Hi. Asterisk doesn't currently support fax pass through as far as I know. W/o fax pass through the faxes don't work well at all. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lach Dunlop Sent: Thursday, March 04, 2004 11:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA 186 cannot make a call Hello, We have a * box up and running with a handful of SNOM 200 phones. It is working very nicely. I am trying to add an analog phone via a Cisco ATA-186 box. The ATA-186 registers fine. It will receive a call. But when it comes to dialing out we get nowhere. :( The ATA-186 is @ firmware level 2.16.2. Are version of * is CVS-022504 and our Linux kernel is 2.4.20 (SUSE 8.2) Our end goal is to attach a fax machines to the ATA-186 Thanks Lach ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186
Cisco ATAs come in two types ATA186-I1 with 600 ohm impedance and ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF in parallel) What is the difference between the two ? Which one is suitable for Europe ? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA 186
Search the list - there's a detailed answer on it. I have two of the I1 version (at least that's what they say they are - ProductId: ATA186I1) and they work with UK spec phones. All you need to watch for is that UK phones are three wire and US phones are 2 wire. Maplin sells an adapter to sort this out (Part no. VD36P). Iain --On Wednesday, February 11, 2004 4:54 pm +0100 Dawid Mielnik [EMAIL PROTECTED] wrote: Cisco ATAs come in two types ATA186-I1 with 600 ohm impedance and ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF in parallel) What is the difference between the two ? Which one is suitable for Europe ? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA 186 / FXO card problem
We had calls dropping every few mins but after we have upgraded ATA's firmware to 2.16 the problem was solved. Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 / FXO card problem
Hello, I've got a Cisco ATA 186 from Vonage plugged into my Asterisk box with a X100P card. This works great for the most part, but I'm having a disconnect supervision problem. I suspect the Cisco device doesn't provide any sort of analog disconnect supervision when it gets a SIP BYE message indicating the far-end has hung up. This causes Asterisk to leave the channel up indefinitely sometimes, if the call was in an app that doesn't time out eventually. Does anybody know of a trick to make either Asterisk deal with the Cisco's lack of disconnect supervision, or a way to make the Cisco ATA-186 provide this signaling? Thanks, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
John, Thanks for the detailed guide. As you mentioned, the situation where two ATAs behind NAT want to establish a direct connection is not resolved yet. In that case, the canreinvite would have to be set to no and some other solution outside of * would have to be used to traverse the NAT. Have you tested any alternatives? Rgds Dan - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 29, 2003 7:35 PM Subject: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what needs to be done to get it working with Asterisk. It's not pretty, it's not HTML, but it's a lot of hints that I've collected from the list and other sources over the last year or so: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
No, I have not tested alternatives. Perhaps Mark can interject a comment here on if he has REINVITEs working for devices behind different NATs or if that is on the agenda? I haven't experimented widely on SIP/NAT interactions since it became stable in the CVS code. JT John, Thanks for the detailed guide. As you mentioned, the situation where two ATAs behind NAT want to establish a direct connection is not resolved yet. In that case, the canreinvite would have to be set to no and some other solution outside of * would have to be used to traverse the NAT. Have you tested any alternatives? Rgds Dan - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 29, 2003 7:35 PM Subject: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what needs to be done to get it working with Asterisk. It's not pretty, it's not HTML, but it's a lot of hints that I've collected from the list and other sources over the last year or so: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
Hi John, Nice job! I just ordered a Cisco ATA186 and I must receive it in about two weeks. This is exactly what I'll need then... I have played some times ago with an ATA186 and Free World Dialup (SIP) and Cisco Call Manager (MGCP) but I have no experience with ATA186 and Asterisk. Thanks, Dan - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 30, 2003 1:35 AM Subject: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what needs to be done to get it working with Asterisk. It's not pretty, it's not HTML, but it's a lot of hints that I've collected from the list and other sources over the last year or so: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
DTMF seems to be out-of-band as a default (or, at least, it's auto-negotiated) and LBRCodec doesn't require mucking with, so I only change the G.711 VAD settings for each channel. More wasteful, but sounds better when you're using cordless phones. JT John's guide goes into a lot more detail, which is nice .. one thing that caught my eye, was the audio mode of 0x00140014 instead of 0x11241124 (as other Asterisk FAQs have suggested). Is there an advantage to your audio mode? Always lookin for better quality .. Thanks, -d [snip] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
