similar to: Peer SIP authentication with Taqua switch

Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN via SIP on a Taqua 7000 switch? My local carrier recently upgraded software and changed their configs so that signalling and media are on different cards (and hence different IP addresses), and it's causing issues. I suspect there are other factors at play... it may or may not be behind a properly configured SBC.

Are there any other Taqua users out there? We have a trunk to a Taqua switch through our ITSP and all outbound calls have the ANI of the primary number on the trunk regardless of what outbound caller-id we generate. This is more than a little annoying, as it interferes with single-number calling, find-me/follow-me, and other features we're using with Asterisk 1.6. Is there anyone with

Anyone have any luck configuring a SIP trunk on a Taqua to talk to Asterisk? We were initially set up as a subscriber (access line) but that had some undesirable side-effects, such as quashing the ANI on outbound calls. Looks like we're going to have to reconfigure the trunk as a "network gateway". I asked their Director of Product Management for product documentation but

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

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

Hi. I have a Supermicro 5018D-FN4T (Xeon D-1541 based SBC) that I use for virtualization. I?m running Centos 7.3 on it (updated), with the CentOS-QEMU-EV.repo repository as the source for virtualization packages. I run an Ubuntu 16.04-2 guest VM on it, which is ordinary enough. What?s perhaps less ordinary is that I?ve attached a Lexar Media, Inc. ?Lexar Professional Workflow CR1 CFast 2.0 USB

Adding Paolo and Miroslav. On Sun, Mar 12, 2017 at 11:41 PM, Philip Prindeville < philipp_subx at redfish-solutions.com> wrote: > Hi. > > I have a Supermicro 5018D-FN4T (Xeon D-1541 based SBC) that I use for > virtualization. I?m running Centos 7.3 on it (updated), with the > CentOS-QEMU-EV.repo repository as the source for virtualization packages. > > I run an Ubuntu

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

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,

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've been using uw-imap for some time on my linux system and have been running into issues with it so I've decided to move to Dovecote, so far it seems to have solved the issues I've been having however I need/want to move the incoming emails out of /var/spool/mail/{user} in the same (or similar fashion) that uw-imap did, and I found the snarf plugin. However whenever I enable the

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

I've now upgraded dovecot from 2.0.21 to 2.1.10 and the good news is I no longer see dovecot crashing when loading the snarf plugin however snarf still does not do anything except make the inbox disappear. I've come to the conclusion that either snarf does not actually work, possible, but I doubt it, or more likely I have a configuration issue preventing it from working. The system is

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

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

I've got a few errors I'm trying to track down, probably all related... Aug 15 14:03:14 xyzzy dovecot: imap-login: Login: user=<jeff>, method=PLAIN, rip=100.8.22.62, lip=132.238.254.34, mpid=4803, TLS, session=<fLvfl85WntFkCBY+> Aug 15 14:03:14 xyzzy dovecot: imap(jeff): Error: namespace configuration error: Duplicate namespace prefix: "" Aug 15 14:03:14 xyzzy

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 =>

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