similar to: lookup for extensions on another SIP Proxy

Hi all, is it possible that asterisk uses two proxies with SRV? The enddevices are registered on one of the two Proxies (Kamailio). The two proxies communicate with each other. And asterisk can choose one of this proxies with SRV. asterisk | \ | \ Proxy1 Proxy2 I have tries to solve this problem with two trunks for this proxies and Dial(... at proxytrunk) but on this way the

Hi, all, I am a beginner of asterisk SIP, now I have 3 pc, one runs asterisk as a SIP proxy, and the other two run softphone(Ubiquity) as User Agents. As below: User Agent <------------> Proxy Server <----------------> User Agent (Ubiquity) Asterisk SIP (Ubiquity) My sip.conf and extensions.conf is as follows: sip.conf [general] port =

Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the

Hi, We''ve an application that uses url_for in controllers and views. In views, url_for generates a relative url (as if :only_path where used). All is fine there. However, in controllers, url_for generates a full url, with the host name. This causes problems when we have a chain of Apache proxy servers: My Browser ---> Proxy 1 ----> Proxy 2 ----> Phusion Deployment Server. In

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

Hi List, I have a szenario where I need to reach a host on the internet from a "firewalled" network but there is a HTTPS-proxy runnnig. As some people know you can tunnel all TCP-connections through this proxy because it can't decide if someone is really doing SSL or just Telnet to port 443 (or use SSH in our case). So I've written a patch for ssh to make it send the CONNECT

Hai,   I noticed something strange in the keytab file on my member server. This is a followup of : [Samba] winbind question. (challenge/response password authentication) Samba 4.5.3 on Debian Jessie.   Leave the domain. net ads leave -k Deleted account for 'PROXY2' in realm 'REALM'   I checked in windows, and the computer is gone in the “Computer” ou.   Removed the

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the

Hai,   Im setting up a new proxy and im testing a bit around. Goal is, get everyting working with minimal changes to the system.   Setup: Debian 8 with NFS nfsv3 and v4 (krb) automounts,  winbind 4.5.3 , squid 3.5.24 (with ssl support) Which is basicly a copy of my other proxy but a new install with more systemd and less packages used.   Working: -          ssh logins with AD users.

Hey there people, i've been scouring the voip-info.org wiki for a few hours and dont seem to be able to do anything in regards to this, so I hope someone can help me here. Now i Have my current configuration as thus.. X-Ten (SIP) -> Asterisk -> Mutliple Peers via SIP or IAX2 Now on one of those SIP Peers id like to call users by their usernames... eg netadmin@some-isp.net, however I

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

Hi, i install the Asterisk PBX on a linux machine with i4l to connect to PSTN (EuroISDN). And i configure a very simple dial plan in extension.conf. After this, i connect with a SIP program to asterisk and would call my cellular phone, but got this error: -- Executing Ringing("SIP/ACzerniak-0904", "") in new stack -- Executing Dial("SIP/ACzerniak-0904",

My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the