similar to: Bridging and unbridging channels

Hi All I facing problem in bridging two SIP channels . I having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error "Bridge stops bridging channels SIP/XXXX-3be3 and SI P/primus-9381"#12539;& call drops. May 13 17:25:19 VERBOSE[7491]: -- SIP/primus-9381 is making progress passing it to

Hi, I am writing this email in sincere hope that somebody experienced similar/same problem and found solution to it. The classical bridge setup could be achieved using brctl tool, which could be found at: http://home.regit.org/br2684.html I have an ADSL CO line card Linux based [2.4.24 kernel] with 8 nas interfaces and eth0. Rather than having bridge, the decision was to have a router, which

Hi All, I am just looking at using Asterisk now and the first thing I need to do is via pass two external numbers to asterisk and call out connecting the calls togther. These will be through our physical PBX connected to the asterisk server. We are essentially trying to connect two external numbers through asterisk with us incurring the cost of the calls. I've been reading on call out

/-----------------------\ | | |eth0 |eth0 |-------| |-------| | |eth1 eth1 | | -------- A |____ _______| B |----- | | \ / | | --------| | | --------| | | | |

Hello, I have an incoming Voip call, where an intro message is played (How can I help you today?) and the answer is recorded - with Record(tst:gsm,3,30). The call is not bridged. I also have an internal SIP phone which dials into an extension as soon as the Voip call lands, and spies on the unbridged Voip call, using ChanSpy. All is well before start of Record () - ChanSpy can hear the

hello everyone! im new here and i just would like to post my problem about the dial application. when i'm calling from an outside line to our pstn which is channel 4 in our digium card. the card recognizes the call and when i dial a number i.e., 1877-LINUX-ME, channel recognizes the call and the call is bridged from 4 to 2, my problem is that the dial application doesnt seem to apply because

(originally filed as a bug in Fedora's bugzilla, see https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=171933) Greetings, Using Fedora Core 4 on a Dell PE 420sc. Malfunctioning kernel is smp-2.6.13-1.1532_FC4. Properly functioning kernels included smp-2.6.12-1.1456_FC4. Network performance is extremely poor when using bridged network interfaces. When not using brctl, the interface

Hii All, VLAN is create in manageable switch, like Cisco Catalyst and other vlan cap= able switch. eac port in that switch can be assigned in different VLAN. so,= that switch capable to insert 802_1q to each incoming frame (Please correc= t if its wrong, i just newbie in VLAN).I want build VLAN but i don't have v= lan capable switch. Is possible to create a vlan capable switch using switc=

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of

Need SIP KEEPALIVES in Asterisk, but QUALIFY won't presently work for you (due to it's channel disabling behavior)? Someone posted on the list that they would like to split "keepalives" and "qualify" into different features. Sounds like a good plan, but until that is done you can turn "qualify=" into a keepalive mechanism, without disabling your channels.

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third

Anil Philip <goodnewsforyou at yahoo.com> wrote: see http://code.google.com/android It would be good of members from here will step in and offer google help to implement speex. Without exaggeration, android will transform the mobile world. Ivo Emanuel Gon?alves <justivo at gmail.com> wrote: On Fri, Apr 11, 2008 at 10:58 PM, Anil Philip wrote: > I recommended in a bug report that

Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking

Hi gang. I am synchronizing Windows machines (cwRsync/cygwin) with FreeBSD server. A batch file is used to initiate the process (shell invoked rsync daemon). My question is whether I need to tell my users to close any affected applications (ex: Outlook; I am synchronizing the outlook.pst file) during the procedure? Peter __________________________________________________ Do You Yahoo!? Tired

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GrandStream102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end

I was incorrect in a previous email... The situation in question is this: Asterisk <---BRI---> PBX <---BRI---> PSTN There are Samsung extensions on the PBX and SIP extensions on Asterisk. I want to be able to use TAPI to initiate dialling, and the PBX has no such feature so Asterisk must initiate it. For an Asterisk initiated call from a PBX extension to a PSTN number, this works as

hello i am a new user for this group. i am working at a ISP. here i want to made a bridge firewall i am using fedora core 3. i want to block a serirs of ip address 192.16.18.0/255.255.255.0 and want to give the accesss only 172.16.18.0/255.255.255.0. but iptables not be able to block ip;s its passes all the ip series. i made my machine as bridge. i think my bridge passes all the

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Hi list, Running Asterisk 1.4.10: When using the M() option for Dial to execute a macro, then executing a Read within the macro, once streaming of the audio file specified in Read has completed, and the channel attempts to read input from the destination channel where the macro is executed, the source channel stops ringing/moh, and audio from the source is bridged into the destination. I have