similar to: problems with asterisk and the IAX protocol

Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: >Pamela, >Did you resolve the problems you described? >I didn't see a reply on the list but I may have missed it. > >-Kevin > >-----Original Message----- >From: Pamela Weis [mailto:peawy@gmx.at] >Sent: Thursday, August 05, 2004

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

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

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

I just setup CENTOS 4.7 with latest patches on DELL server. I also configured NTP point to out time server. I found /var/log/messages file every 20 to 30 minutes will generate a error message : Mar 15 14:28:15 SER1 ntpd[25037]: sendto(172.29.21.16): Invalid argument Mar 15 14:45:22 SER1 ntpd[25037]: sendto(172.29.21.16): Invalid argument Mar 15 15:02:29 SER1 ntpd[25037]: sendto(172.29.21.16):

Hi all, I have a time series that contains double seasonal components (48 and 336) and I would like to decompose the series into the following time series components (trend, seasonal component 1, seasonal component 2 and irregular component). As far as I know, the STL procedure for decomposing a series in R only allows one seasonal component, so I have tried decomposing the series twice. First,

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is 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

Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded 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

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

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 that the RTP don?t go

Hi all. Ive recently setup two Asterisk boxes (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

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

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

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where 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

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 register

Hi, I am using iax2 trunks between asterisk servers and am having a callerid problem. We are using realtime sip clients distributed between multiple servers. Only in test now but have run into a calleeid problem - the name of the called party is not displayed if the called party is on a different server, it works if the called party is on the same server. On each server sip clients show calleeid

Hi, I have configured the cdr throught ODBC with this files: /etc/cdr_odbc.conf [global] dsn=asterisk2 ;loguniqueid=yes dispositionstring=yes table=cdr ;"cdr" is default table name usegmtime=no ; set to "yes" to log in GMT hrtime=yes ;Enables microsecond accuracy with the billsec and duration fields /etc/cdr.conf [general] enable=yes unanswered =