similar to: VoIP Provider problems

I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this

I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change. Inbound calling has been down for 2 days. Beyond the "We are currently experiencing in-bound call issues with a carrier partner in some areas. We are aware of the issue and our engineers are working to have it resolved as soon as possible" mantra their

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Does anyone know of a good VOIP dialtone provider in the northern Chicago area. My client has tried Broadvoice and Mix and is having problems with latency in the middle of the traceroute between him and the provider. Thanks, Greg No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date: 1/4/2008 12:05 PM

Hello, I''ve got this problem. There is an linux server with 2.4.24 kernel and pinging from him to internet (or from lan) ping randomly jumps up: 64 bytes from fortas.ktu.lt (193.219.160.131): icmp_seq=387 ttl=59 time=30.0 ms 64 bytes from fortas.ktu.lt (193.219.160.131): icmp_seq=388 ttl=59 time=32.6 ms 64 bytes from fortas.ktu.lt (193.219.160.131): icmp_seq=389 ttl=59 time=34.9 ms

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

Hello Tom. Sorry, don't answer on my previouse letter, i forget to set subject. I fix this in current. And now about my question. I ask you before about method of stopping RFC1918 traffic on external interface and you advised me follow rule: REJECT! all net:$RFC1918_NETS Can i replace this rule by 'norfc1918' option in 'interfaces' file for this interface?

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried

Stephen Clark wrote: > Robert Noland wrote: >> On Thu, 2008-11-13 at 07:48 -0500, Stephen Clark wrote: >>> Julian Elischer wrote: >>>> Stephen Clark wrote: >>>>> Julian Elischer wrote: >>>>>> you will need to define the setup and question better. >>>> thanks.. cleaning it up a bit more... >>>> >>>>

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

I have a shorewall 2.0.14 running on a single interface machine (nwww in the log below) that is attempting to be well screwed down. The policy file reads:- #SOURCE DEST POLICY LOG LEVEL LIMIT:BURST fw net DROP info net all DROP info # The FOLLOWING POLICY MUST BE LAST all all

Hello, i have a little question. My system: ip route: 0.0.0.0/1 via 10.8.0.5 dev tun0 default via 192.168.2.1 dev br0 proto static metric 425 10.8.0.1 via 10.8.0.5 dev tun0 10.8.0.5 dev tun0 proto kernel scope link src 10.8.0.6 88.198.140.127 via 192.168.2.1 dev br0 192.168.2.0/24 dev br0 proto kernel scope link src 192.168.2.101 metric 425 192.168.122.0/24 dev virbr0 proto kernel

hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier.

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Is there a traceroute equivalent in the VoIP world? I would like to see the route a call takes after it gets to the gateway. Basically showing all the hops until it reaches it's destination or PSTN termination. -Dave

Hello, I have two broadvoice accounts. Lately, very often my broadvoice accounts are in unregistered state. When I log into asterisk I see: voip*CLI> sip show registry Host Username Refresh State sip.broadvoice.com:5060 yyy@s 120 Request Sent sip.broadvoice.com:5060 xxx@s 120 Request Sent xxx and yyy are my ohone numbers, when I do tcpdump on my router, I see only packets comming out my

What can I do to UNDERSTAND why, periodically, I can't reach www.centos.org (but everyone else can)? Every few months, http://www.centos.org just drops off for me and for me only. It's not the web site, because I can reach it via a proxy (e.g., TOR browser bundle) and other tests show it to be alive (e.g., http://www.downforeveryoneorjustme.com/centos.org or

hi everyone I'm having a bit of a puzzle on one centos system, I have four interfaces: nm-team1 10.5.6.100 em1 192.168.2.17 em2 192.168.2.18 p3p3 172.25.12.222 I do: ping 10.5.6.17 -I em1 PING 10.5.6.17 (10.5.6.17) from 192.168.2.17 em1: 56(84) bytes of data. 64 bytes from 10.5.6.17: icmp_seq=1 ttl=64 time=0.367 ms the same for -I em2, for -I em1 and naturally for -I nm-team1 - all good.

Hi, how can ip instances be used with vlans ? Is that possible ? I don''t know how to configure this, if I only have one interface (e.g. bge0), and want to assign e.g. bge1000 to a zone. Any cookbooks available ? Thanks Detlef