similar to: outgoing sip calls work; incoming calls fail

Hi, My setup is : PSTN --------- ISTP Network ----------- Router ------------- Asterisk ---------- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I think I should have those phones directly exchanging RTP data

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

same problem with Windows XP SP3 some problem accessing with XenCenter from different clients (XP and W7) the only interesting logs seem: [20130906T07:25:14.943Z|debug|titanio|191 INET 127.0.0.1:80|Connection to VM console R:73f4468306ca|console] Proxy exited [20130906T07:25:14.947Z|debug|titanio|96 INET 127.0.0.1:80|event.from D: 1327d493dfe8|backgroundscheduler] Removing function

Hi, I've noticed that asterisk seems to stop relaying DTMFs after a call has been up for a while (~10 mins). I was just wondering whether this was intentional, or a bug. In detail here's my setup SIP Gateway --> Asterisk --> E1 --> Asterisk --> SIP Gateway The LHS gateway sends RFC2833 DTMF messages to the first Asterisk which bridges them onto the E1. They then get

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Greetings all, We have had one of our mail servers running 4.8-STABLE reboot every few hours The following messages were found in the /var/log/messages >Apr 13 13:19:20 ms /kernel: panic: malloc: wrong bucket >Apr 13 13:19:20 ms /kernel: >Apr 13 13:19:20 ms /kernel: syncing disks... 75 27 27 27 27 27 27 27 41 3 >3 3 3 3 3 3 8 3 3 3 3 3 3 3 6 3 3 3 3 3 3 3 3 3 3 3 3 3 3 >

On 2014/7/3 11:20, Ilia Mirkin wrote: > Hello, > > A user (cc'd) reported that nouveau's enabling of MSI causes the card > to not work on his setup [1]. I think the situation is that MSI is > just not supported by the underlying motherboard, even though the > card, and probably bridge, support it just fine. It's a very old > board. The nouveau code does: What is

Hello, I'm trying to write a Vorbis decoder (just as a hobby) but I've got a problem with reading the setup header. According to the Vorbis documentation the first data that should be read after the initial 'vorbis' string is an eight bit integer which when incremented by one, contains the number of codebooks in the header. My problem is that whenever I read this number it seems

Hello, A user (cc'd) reported that nouveau's enabling of MSI causes the card to not work on his setup [1]. I think the situation is that MSI is just not supported by the underlying motherboard, even though the card, and probably bridge, support it just fine. It's a very old board. The nouveau code does: pmc->use_msi = pci_enable_msi(device->pdev) == 0; Does it need to do more

Hi everybody, Is someone can confirm me that there are 2 bugs never fixed: - first in the stat command. Only with the -x option. If you execute stat -x on /tmp or /usr/bin/passwd parameters for example, the numeric representation of mode is wrong. The "special" bits are always 0. No suid-bit, no sticky bit! - Second. Because of a missing suid-bit on the newgrp command, this

The Asterisk Development Team has announced the release of Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to an IAX2 denial of service vulnerability. For more information about the details of this vulnerability, please read the security

The Asterisk Development Team has announced the release of Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to an IAX2 denial of service vulnerability. For more information about the details of this vulnerability, please read the security

I see a lot of these on an otherwise idle Asterisk 1.6.0.15: Extension Changed 773[Hints] new state Ringing for Notify User 792-00041327d17e-1. Then a little while later it changes to InUse or Idle, completely randomly. It happens for many different combinations of phones and watchers. There are no calls being made, so I can think of no reason why this happens. Obviously it wreaks havoc with

Good afternoon, A strange one - library(rJava) works for root but not for user sean On 64-bit Ubuntu 9.10, using R as root (sudo -i), I can say library(rJava) and it works fine with no error message of any kind. But logged in as myself user "sean", it can't start... sean at SeansPC:~$ R --no-save --vanilla R version 2.10.1 (2009-12-14) Copyright (C) 2009 The R Foundation for

Hi, I recently upgraded our asterisk server from some 1.4 version to version 1.6.0.15. From this point on my AGI scripts aren't working anymore, here is a simple example: [isdin] exten => 83086921,1,AGI(test.php) exten => 83086921,2,NOOP("MARKE1") exten => 83086921,3,WAIT(2) exten => 83086921,4,Hangup() /var/lib/asterisk/agi-bin/test.php

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me

My set up is 1.6.0.15 with the digium fax modules. I want to capture a fax from the internal analog fax machine (using an SPA2102), and then resend it. I know the internal extension of the fax machine, and for now I'm just testing it to one outside fax machine if I dial 8447. In particular, I'm completely unfamiliar with the use of "G" in the Dial app. exten =>

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing