similar to: innomedia ATA's

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

All, I'm having a problem with an Innomedia 6328-2Re (old Sunrocket Gizmo). It keeps losing registration after a period of time ranging from a few minutes to a few hours. It seems that right before it loses registration, it fails to send a second register (after the 401 unauthorized). Here's a transcript from wireshark (at the end). The last message is all that's received and

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had

Hi, Does anybody have experience working with Aculab groomer II, to convert between ISDN E1 and non-ISDN T1, or anything similar. I am looking for sample config files. We have asterisk as ISDN E1, but for testing we set it up as regular T1 if we get sample config files. Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Hello list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran, dealt with them in the past, won't deal again.) I talked to Digium and the

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

Hello list, For a customer I need to detect blue alarms on his T1 trunks. His server is in running asterisk and zaptel 1.4. I have a tool to generate all sorts of alarms, but on generating blue alarm, zaptel recognizes them as red alarms. This is not good for us as we need to get blue alarm as blue alarm. I checked using: cat /proc/zaptel/* zttool Are these versions of asterisk and zaptel

your sip communication altogether. Have you tried changing IP address of your asterisk server? If changing IP works, then probably your provider has blocked you sip communication by IP only. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-23 7:22 AM, "bilal ghayyad" <bilmar_gh at yahoo.com> wrote: Hi All; I have my friend that use his mobile (Nimbuz) to connect for the

Hello list, I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran) Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part -------------- An HTML attachment was

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]