similar to: No audio on remote SIP calls

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

hello, i'm having a wierd problem with 0.31 tftpd-hpa. i'm using xinetd, with this config: service tftp { disable = no socket_type = dgram wait = yes user = root log_on_failure += USERID bind = 10.13.0.254 server = /usr/sbin/in.tftpd

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Has anyone had experience using a SIP trunk provided by Paetec over MPLS? With or without FreePBX Regards, Jared Baxley -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120926/436ee928/attachment.htm>

Hi I;ve trixbox installed with 2 NICs. One NIC carries the MPLS-VPN traffic (only a 1 MB link without internet for carrying voice to another site) while the other NIC has a connection with public IP for internet services on that machine. the first NIC (eth0) has IP of 172.16.0.1 and is connected to router with WAN IP: 10.18.6.254 , the second IP is 203.234.82.98 (eth1). i want to have the

Good afteroon all, First of all: thanks for everybody who is willing to think this through with me: I'm having some issues regarding call quality between some calls. Let me try to explain the situation first We have a Asterisk 11.16 server based on the Xivo distribution. There are 2 servers running in cluster (Active Passive), both virtual with the following config: Quadcore CPU 8 GB ram

Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk

We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice,

Hi, This question is both technical and business related. I've got a prospective customer in France which belongs to Hotel industry. He has a lot of visitors coming from the Nederlands. I'm studying the opportunity to offer phone services to those visitors. The service I'm thinking about is plain local call termination : hotel guests cost effectively call their relatives in their

Hello, I am having a problem with my current SIP over VPN setup. We have a server running asterisk at our office. All the phones in the office are on the same network / local to this server. We also have two employees with home offices using SIP phones over VPN to connect to the asterisk server. These phones have no problem with calls to the phones in the office, however there is no audio

I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

Hello, sorry for this little OT post. In my company, we have 2 distant facilities, with people at each facility working on the same files. The 2 facilities are connected through MPLS with about 10MBytes/s BW. Saving the work files to servers located at one or the other facility has became a pain for the people accessing the files from the remote site. Trying to improve files

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

On 2018-09-09 10:27, Antony Stone wrote: <snip > 1. Try removing one of the two commas. > > 2. Take a copy of your dialplan, and then strip out *everything* except > the > one context and the one number you want to match: > > [0705680837] > exten => 31705680837,1,NooP( Incoming 31705680837 on CC) > same => n,Answer(); > same =>