similar to: Outgoing SIP to SER causes LOOP BACK message

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

I'm having trouble with both my Polycom IP600 and IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a phone call that lasted 1hr 39mins which is my record so far. I cannot figure out what is causing the call to terminate and I am hoping somone on this list can help me. In this example both the phone and the asterisk server have public IP addresses so NAT shoul not

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

What is the relationship between the peer definitions and the register command? In reviewing the sample sip.conf it seems that you can place the "sip_proxY" peer as the hostname. Is this correct? This question adds the the Broadvoice thread and where to place the dtmfmode variable. sip.conf --- (asterisk sample) -------------------------------- ;register =>

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

Hi, > >... > >drivers/isdn/capi/kcapi.c:1014:47: error: macro "INIT_WORK" passed 3 > >arguments, but takes just 2 > >make[2]: *** [drivers/isdn/capi/kcapi.o] Error 1 > >make[1]: *** [drivers/isdn/capi] Error 2 > >make: *** [_module_drivers/isdn] Error 2 > >#+ LOG INFO: pwd:/usr/lib/eicon/divas/src > >#! LOG ABORT EXECUTION DUE TO

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

Hi; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter

Hello, I have installed an asterisk server with 7960 sip phones and an eicon card (diva 4bri server, I use only three channel) using capi channel. Everything seems to work fine, but from time to time (two or twice a day) during a call I get a capi error in asterisk log and asterisk doesn't work anymore: ==== Apr 1 10:38:24 ERROR[20495]: File chan_capi.c, Line 738 (capi_write): error

Hello, I want to use a 4 port ISDN card from EICON/DIALOGIC in our asterisk server. The system is a Ubuntu 7.10 with kernel 2.6.23.1. The compilation of the kernel finishes without any problems. I have downloaded and installed the deb-source package that EICON/DIALOGIC offers. Th installation script crashes with the following error messages: # LOG ---- START SECTION read kernel version