similar to: SIP registration problem

We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to 999@pbx.itsp.com REGISTER attempt 2 to 999@pbx.itsp.com Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why

I'm being told by my ITSP that my Asterisk shouldn't be challenging their system to authenticate (i.e. a 401 response) when they send me a SIP MESSAGE (or I suppose a SIP INVITE for that matter). But I'm not sure what a pjsip.conf configuration for that looks like. How does one associate an incoming call/message with an existing authenticated outgoing registration so that Asterisk

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

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

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

I have a new Asterisk system going into production next week and I'm a bit stumped as to the best way to handle the Dialplans for it. The Asterisk system is replacing 4 separate PSTN lines with both SIP & PSTN inputs. The setting up of the dial plan is giving me some design headaches, which probably means I'm missing something obvious and doing this the hard way. I have separate

Hello, I'm trunking with an ITSP that, when treating an outbound to an unknown destination, either: - send a SIP error code (I can't be more explicit, at the moment), - or cast a pre-recorded audio message using Early Media. At the same time, I'm also trunking with Contact Center solution which doesn't support Early Media. Beside asking my ITSP to treat calls consistently or

Great ! I'm gonna it try ASAP ! Is there another way (ie not using different ports) to get several trunks to a given ITSP ? Let me explain this a bit further. My setup is: ITSP <---- SIP----> Asterisk <----> Phones For various reasons, I want my Asterisk box to have several trunks/SIP account with my ITSP. First method, is to configure a specific port for each trunk: ITSP will

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

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

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > you can try line functionality on the outbound registration which > may or may not work[2] (requires the upstream to adhere to the RFC, > which not all do). My provider seems to implement this. However even with the line=... in the: SIP to address: sip:5555551212@<my_IP_address>:5060;line=dpnlyiu res_pjsip is still

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

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 folks I have a handset talking to Asterisk, which in turn puts the call through to an ITSP. The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP). The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset,

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