similar to: Problem with NAT settings?: SIP reply CALL-ID from ITSP has internal address in host part

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. ?I can

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

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

To all those that wanted to know how I was filtering particular ICMP packets here is a few snippets from my firewall script which is based on one by Ian Hall-Beyer. I hope this helps you get started. Also note the output of the command: ipchains -h icmp Shawn Mitchell mentioned blocking all ICMP echos and especially broadcast echos. Perhaps he''d care to elaborate with a similar

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

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 All, When I want use Asterisk as a PBX to cooperate SIP ITSP, I can not set the caller ID, so SIP ITSP do not accept the call. In Asterisk, I set a account in sip.conf to register on ITSP SIP Server: register => 6292@218.1.121.237/6292 And I added a user 6292 in Asterisk just like the account on ITSP SIP Server: [6291] type=friend username=6291 callerid=6291 host=dynamic

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is

We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

There are a number of them, try Comwave, Voxip or Wiztel. Depends on what you need we may also provide it... email me privately if you're interested. Some provide IAX, some only SIP, H323, & MGCP... -----Original Message----- From: hugolivude [mailto:hugolivude@gmail.com] Sent: Thursday, February 02, 2006 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

I am using voip with Vodafone as SIP peer for outbound telephony and i have a huge problem establishing calls to other people. It works like in 1 of 5 tries. The peer is sending SIP 480 temporarily not available. It took a while to identify this, because on the phone you just hear busy tone. On inbound calls i have not detected problems yet. Calling to mobile numbers works better than to

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at