similar to: question about zttest

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

I have some DID numbers that come into Asterisk via PRI, then connect to a Panasonic DBS PBX via PRI. Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's PRI. The sync LEDs on my PBX show that it is synced to Asterisk via the PRI. I have users complaining about random hangups. What is the best way toi approach finding the issue? ref: zttest looks good. ./zttest

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

I get the following message on a Centos 5 system (really a Trixbox 2.4 build on Centos 5): Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled What does this mean? This message occurs about 30 times/sec for about 45 sec. Then my Bluetooth token starts up. Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled Jan 14 00:13:00 sip2

Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco.

Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ?anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best:

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.000000 -- Worst: 99.780273 -- Average: 99.975763

Hi Francois, Thank you for your interest. I tried the card alone so I don't think is an IRQ problem (anyway is there a way to be sure?) To be sincere, in all systems I saw (even working ones) there is an IRQ balance failed where Linux boots. The system is new, I tried different processors (P4 and Celeron) with SMP kernel and without. The card has the echo module, could not be

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours