similar to: SIP message 183 and in band info

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are

Hello, I am trying to setup a2billing system for asterisk. I have installed it corectly, but I have not found any users manual. I do not understand the whole structure. How do the parts like calling cards and sip friends cooperate together? I simply need to know how to make a call through it. With all the features like CDR's, etc. Can anybody help me with this? Thanks in advance.

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hello, I've got such a problem. I'm configuring Asterisk as a backup server, if call to the first one fails. My problem is, that the redirection from the sending machine work so, that in the INVITE line of the INVITE message is the presentation number of the Asterisk server and in the To line is the real called number. So I need to setup Asterisk so, that it will ignore the number in

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

Mandi! Rowland penny via samba In chel di` si favelave... > yes, Provided you use the right attribute to search on ;-) Ah! ;-) Just i'm here, i test three condition in account flags, eg: UAC=$(ldbsearch ${LDB_OPTS} -b "${BASEDN}" "(&(objectClass=user)(sAMAccountName=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-)

I have a Steam version of DAO + Awakening and have had no troubles with it up until now when I've bought some DLCs from official bioware store. For any DLC I bought I'm getting "Unable to load area" error right after character creation or importing (or, in case of "Leliana's song" --- right after "Play" is pressed in "Other campaigns" menu). Its

I hopefully cleared all SAMBA files and set up a fresh ADC using: samba-tool domain provision --use-rfc2307 --domain=UAC --realm=UAC.MGR --server-role=dc --dns-backend=SAMBA_INTERNAL --targetdir=/srv/files --adminpass="secret" --option="dns forwarder=172.16.6.11" The provisioning seemed okay, i.e. nothing hints at any errors and I see a DOMAIN SID as the final entry as

I need to do some testing, but before to hit by head on a known wall, i ask here. My AD domain get used (via PAM/Winbind) to give access to some other dervice, most notably here dovecot. When password expire (or users change it) the MUA try the old password some times, then ask for a new password; users cleraly get scared, press randomly 'OK' or 'Cancel', but if they press 2-3

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hi All, Excited to see mention of T.38 passthru with 1.4.0.... BUT it seems like "Emperor's New Clothes" from my perspective. As I understand it, T.38 passthru only works via a re-invite mechanism - so two T.38 devices try talking fax and request a T.38 media stream. Asterisk is now savvy enough to realise it can't do it, so opts out - "Hey Guys, I don't know

Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hai, small note here.. > > Setting up the Admin user is interesting. I will try that. > But I could as well add myself to the domain admins group. > The name is arbitrary. Yes the name is but now you are working with Admin rights. Never ever do that. Please dont, use the UAC. It's so easy to get infected if you work as admin so dont.. Personaly here, at the office, i