search for: asterisk2

Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone on the other side rings. But whenever I pick up the call, asterisk2 hangs up without much warning and then the telephone rings unexpectedly again and again. He...

...X protocol > > >Hello group, > >I wanted to try out the asterisk iax protocol between two asterisk >machines but have several problems with it. >My scenario looks like follows. I am using asterisk 0.9.0 on both machines. > >SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 > >Both SER and asterisk run on a machine with a public IP address. When >the telephone on one side makes a call the telephone on the other side >rings. But whenever I pick up the call, asterisk2 hangs up without much >warning and then the telephone rings unexpectedly...

...s the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > asterisk2 server Now, let me explain the symptoms. d-channel errors on the asterisk1 server on span1 (which is the line coming from the echo cancel from the ISP). asterisk2 server isn't being used as far as I can tell. I've got a red alarm on the port on asterisk1 that asterisk2 is plugged into. f...

...cense' for details. > ========================================================================= > == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf > Found > Connected to Asterisk 1.4.20-1 RPM by vc-rpms at voipconsulting.nl currently running on asterisk2 (pid = 2336) > Verbosity is at least 5 > Core debug is at least 1 3. spa-3102 details: ============= > Product Name: SPA-3102 > Software Version: 5.1.10(GW) > Hardware Version: 1.4.5(a) > LAN IP address: 192.168.0.10 > LAN subnet mask: 255.255.255.0 > > PSTN Line -&gt...

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there is one way voice whereby the B party cannot hear the A party, ho...

...GSI 17 (level, low) -> IRQ 17 in asterisk cli: Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI> transcoder show 0/0 encoders/decoders of 92 channels are in use. I am trying to do 2 experiments: 1) asterisk1 --->g729--> asterisk2 (with transcoding card) Playfile in ulaw format 2) asterisk2 --->ulaw--> asterisk2 (with transcoding card) Playfile in g729 format In the first case I get all calls proceeding and in asteris2 cli Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is...

...re must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the right run, the callers must point to the asterisk2.... Is there some *.config file where I can put my vrid IP, so in automatic the asterisk1 and asterisk2 translate their IP to the vrid? The vrrp is right like I set it.Asterik1 is the master with 192.160.252.1 IP and vrid like 192.160.252.10 asterisk2 is the slave with 192.160.252.2.Via Ethreal the...

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to...

Hi, I need some information on DID and Callback. Please read-on: Question on DID (User1 Calling User2 via normal Telephone line and sending its CLI: Connectivity is as below: User1 ==PSTN==> DigiumE1/Asterisk1 ==INTERNET==> DigiumE1/Asterisk2 ==PSTN==> User2 1. Can User1 make a single stage call to User2 via Asterisk1? Currently User1 is able call User2 on Two Stage basis (Asterisk answers, and then user1 hears a message, and then he is allowed to call) Question on CALLBACK. Connectivity diagram is as below: User1 ==PSTN==> D...

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

...from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at asterisk2 ~> crm_mon -1 ============ Last updated: Thu Mar 28 06:09:54 2013 Stack: Heartbeat Current DC: asterisk2 (b966dfa2-5973-4dfc-96ba-b2d38319c174) - partition with quorum Version: 1.0.12-unknown 2 Nodes configured, unknown expected votes 1 Resources configured. ============ Online: [ ast...

...ATA 2 | +-------+ +-------+ / \ / \ / \ / \ 21 22 10 11 That is, I have 2 asterisks connected via SIP, two ATAs with two lines, and the ATA1 is registered with asterisk1 and ATA2 is registered with asterisk2, and all incoming calls in asterisk2 from the asterisk1 (via SIP), are answered by a DISA. I can make calls between ATA1 and ATA2 without problems (the call will be routed to the asterisk1 to asterisk2, falls in DISA and I call one of the phones ATA2). I am now trying to make the call coming from,...

Hello, I have the fowling scenario: fxs[asterisk1]-----iax-----[asterisk2]e1----e&m---PSTN I want to know the steps to transmit fax from a machine connected to the fxs to a fax machine on the PSTN. The same for dial-up's. Is it possible only with a/ulaw ? What configs I need in asterisk1? Thanks in advance Eduardo

...xes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them. So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two. I have named each box asterisk1 and asterisk2. Does anyone have some working SIP and/or IAX trunk configurations they can send to me? Here is my current SIP config which doesnt seem to work: sip.conf on asterisk1: register=ast1:****@x.x.x.x [100] username=100 type=friend secret=**** record_out=Never record_in=Never qualify=no port=5060 ho...

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to client2 But, in the asterisk2 CLI, I have the error, and with a tcpdump capture, I see th...

Hello, My network is: Client_SS7_1-- -----------asterisk1------asterisk2 Client_SS7_2-- ? I receive a fax from Client_SS7_1 ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Then, asterisk2 forward the fax to Client_SS7_2 I want that the SIP signaling go to asterisk2, But, I need tha...

Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Asterisk2 comes in, it is correctly entering the above remote context and an exte...

Hi, I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in mind, (1)I can not send DTMF tones as Conference...

...it's not supposed to do ? Or is it really a bug ? Has anybody already heard of this bug ? Or does somebody knows another way to achieve the same functionnality (3 way calling with two ingoing calls) ? Thanks in advance. J-M HEITZ & LM Linux Distribution : Ubuntu edgy Kernel : Linux asterisk2 2.6.17-10-server #2 SMP Fri Oct 13 18:47:26 UTC 2006 i686 GNU/Linux Zaptel version : Apr 20 10:18:48 asterisk2 kernel: [44924783.590000] Zapata Telephony Interface Registered on major 196 Apr 20 10:18:48 asterisk2 kernel: [44924783.590000] Zaptel Version: SVN-trunk-r2396 Echo Canceller: MG2...

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN in Asterisk1 and forward the call to Asterisk2 (which is connected to a Zhone 100 channel ba...