similar to: Problem with SIP 480 from ITSP

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

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

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

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

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

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"

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:

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

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

Hello, I want originate a call to some destination, and when B side answes to play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP header to Invite, that's why I'm using Local Channel. This is my extension.ael: context autodialer-local { _X. => { SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

Is there some concensus on where to set callerid when making outgoing calls via an ITSP over IAX? Is this best accomplished in IAX.CONF or EXTENSIONS.CONF? Also, tech support at one ITSP told me that the SetCIDName command doesn't do anything. Is this something that might be unique to their server? Or a general statement of fact? Thanks, Michael -- Michael Graves

SQL help needed.. If I have a MySQL table with dialing codes and a corresponding description (see below) and I want to lookup the best match for a phone number.. What would the SQL look like to do it? or would it take more than just SQL to get to the best result? Thanks.. Later.. Example numbers, (random end digits so I don't know who's they are.) 00442085673456 - UK London

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

https://issues.asterisk.org/jira/browse/ASTERISK-20841 The patch was already posted by someone but then was deleted because of guide lines. Is it really that hard to fix? Since 1.8 there is this problem but nobody seems to care about. Asterisk isnt only used with itsp who dont care about fromdomain. Or are the developers saying, we dont care about people who are using Asterisk in smaller