similar to: Hairpinning problems using IAX2 and SIP

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

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

Hi, My setup is : PSTN --------- ISTP Network ----------- Router ------------- Asterisk ---------- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I think I should have those phones directly exchanging RTP data

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

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

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

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 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

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Hi all, I have the following problem, working with a SIP provider, if i setup my SJPhone to register directly to their STUN server and working over a 384/128 ADSL i have a really good quality, but then if i configure Asterisk to register to the same provider over the same 384/128 circuit the quality is REALLY BAD. The obvious difference is that using directly the SJPhone i am using STUN, while

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

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

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,

Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing

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

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 am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38