similar to: No 2nd invite from asterisk after challenging original invite

In an attempt to connect our Asterisk 1.6 phone system with another phone system called "Broadsmart", they gave me credentials to register to. Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365) watermelon*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time {broadsmart_ip}:5060 N

Asterisk Project Security Advisory – AST-2020-002 Product Asterisk Summary Outbound INVITE loop on challenge with different nonce. Nature of Advisory Denial of Service Susceptibility Remote

Hi all, Today I've stumbled upon a very strange behaviour with an analog fxs/fxo gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html) connected to a CVS HEAD(from today) Asterisk server. This manifested itself after enabling the CallerID on the pstn lines connected to the FXO ports of the module. Both FXO modules have their own sip username/passwords and are registered to the

I recently posted regarding problems configuring Samba. These have all been fixed but I am now struggling with the point and print facility. I have attempted to upload the drivers from Windows XP workstations but this seems to do nothing. So plan B was to manually copy the files to the appropriate print$ share folder, so far so good. The problem I now have is that the rpcclient utility simply

As an updatee to my last post, things are still not working! The drivers did get added but I'm still not sure whether I achieved this via the Add Printer Wizard or despite error messages the rpcclient adddriver did work. I did wonder if the lsa_io_sec_qos: length c does not match size 8 error is because the printer name is too long, especially as when I tried a shorter name such as HP2300 -

Hi, I have used asterisk 1.4.17 to interwork with a SIP Proxy. Asterisk acts as a UAC and registers three users with the SIP proxy. It handles the proxy authentication for register requests well and all three users get registered with the SIP Proxy. However when the Proxy challenges the INVITE sent by asterisk, asterisk is unable to use the credentials for each of those users. If I use

Further to my last post I decided to BUY a copy Suse 9.1 including Samba 3. Not only was Suse very easy to setup I managed to get Samba up and running without too many problems and can now print from Windows clients via Samba. My only remaining challenge is setting up point and print. Thanks for the responses. Regards, Chris Christopher Moss Murray McIntosh O'Brien Wellesley House 204

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

On Sunday 25 October 2020 at 16:27:00, Antony Stone wrote: > Hi. > > I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and > for some reason it's simply not doing it. I've made a bit of progress - I can now get it to authenticate, although it's still not dialling on to the correct number. > I've even resorted to reading the source code

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Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting each others IP 100%. But the PBX02 challenges PBX01's requests all the time even though

I have been unable to configure a Samba server I am testing to enumerate the users and groups in our local NT domain, but I have been able to configure it to enumerate the users and groups in our W2K domain. I am hoping someone has some suggestions for what to try next. The Samba server is running Linux installed with the XFS RedHat 7.2 installer CD. I get the same results running Samba 2.2.3a

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

Hi, according to the Facebook docs, you can add a uid to the fb_request_form_submit button which will pre select the user for the form. So you can do: <fb:request-form action="/my_tasks" method="POST" invite="true" type="MyApp" content="wants to invite you to xyz app"> <fb:request-form-submit uid="FRIENDID" />

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hallo, As part of building a debian package for my use of tinc 1.1 I run the "make check" testsuites. The current git breaks in invite-join.test: FAIL: invite-join ================= + ../src/tinc --config=/home/haegar/tinc/tinc/test/./invite-join.test.1 --pidfile=/home/haegar/tinc/tinc/test/./invite-join.test.1/pid Generating 2048 bits keys: ...........................+++ p ....+++

Hi, I'm installing asterisk 1.0.5 for a partner in China. Since the ping round trip takes typically 600 msec, I doubt, whether voice quality will we satisfying, but that is currently not my concern. The problem is, that most SIP phones or software (e.g. SJPhone) do resend the invite request, after approx 500 msec (measured by ethereal). chan_sip from asterisk seems to have a special

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

I don't understand what a SIP invite is. Certainly it's explained as: "This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between 4 and 6 header entries with contact information inside them." http://www.3cx.com/blog/voip-howto/sip-invite-header-fields/ The article enumerates the headers

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from