similar to: Multiple bridge attempts

Hi all, I came across a new(to me that is) issue. I want to know from others what they have done to resolve this. I have a 4 port digium card with FXO's, and connected to each FXO is a premicell. When I dial the premicell, after about two seconds is says 'ZAP/1 answered', then it takes a few more seconds for the call to hit the cellular network, before the cellphone starts to ring.

Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread:

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com X-Spam-Status: No My third try, humph! Yusuf wrote: > Hi, > > I am trying to understand the RTCP stats in Asterisk. > > 1. I am using the 'h' exten to store the RTCP records in

Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 <- 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]:

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

I fear that list digest did not forward to me all the messages... buying cell phone adapters is quite unfeasible at this point, since the installation at hand uses 8 BRI for outgoing calls, and the customer negotiated very special rates for handling all the traffic through his voice carrier. Moreover, in italy you have 4 cell phone operators, and you should add a bunch of call phone adapters for

Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? -- thanks, Yusuf

Hey all, I have read on voip-info.org that to configure MoH asterisk requires the use of mpg123. I have installed mpg123 and restarted asterisk. But, when i put a call on hold i get this error: May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865 local_ast_moh_start: No class: default Can you help, Thanks yusuf

Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

In article <1251.165.146.69.140.1144596935.squirrel@www.ecntelecoms.com>, yusuf@ecntelecoms.com says... > Hi, > > I have had the exact same problem last week. I have not yet solved it. > So instead I am using ooh323, but would prefer to use oh323. Can anyone > help? I'm glad that I'm not the only one :)) Hopefully we'll find solution to this problem. --

Hi, sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack

Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc..... Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.

I had sent email to Andrea asking him if his blkdev-in-pagecache might cause the recent reports about ulimits being applied to block devices. This is his response which he asked me to forward to ext3-users since he can't post being a non-subscriber BTW, Redhat should whitelist various kernel hackers email address on their mailserver so that they can post freely to various lists Regards,

postfix-users seems to be a subscription only list. I'd recommend incorporating TDMA <http://tdma.sf.net/> to allow for easy discussion by outsiders Andrew Morton (of ext3 fame) had sent this message to postfix-users list I am forwarding so that Wietse can hopefully provide the definitive answer to the question Andrew Morton/Stephen Tweedie seek ----- Forwarded message from Andrew

Hi, my name is Yusuf, I just join with this groups. I have using samba PDC with LDAP as backend. I have a problem to change user password from web. I tried using sudo smbldap-passwd, change permission every file so apache can read / execute that file, but I'm still can't change the user password. Is there any way to change the password only with change the LDAP password (using

Hi, sometimes on by Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack