similar to: SIP Debug information

Hello all, I need to configure a new provider to complete calls to us, the provider gave to me 2 different ip address, one is the default host and another to RTP server, so far as i knew the rtp server should be the same address but different ports, anyway i think i?m completelly wrong about it.. someone could tell me how can i configure in asterisk this connection in sip.conf? Thanks, Chet

Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the "insecure" option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username.

? HI ? I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) ? Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg ? Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858

During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They

Hi Everyone, In order to make life easier and to do debugging easier I want to observe "sip set debug originator" and "sip set debug terminator" on two different putty screens. Trick is that originator calls the terminator. I can of course put two separate calls and get sip debugs at different times but that's not what I want to do. I want both to spit out on my two

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I was actually trying to use the dialplan application that uses 'Dial' and when the: Dial(SIP/XXXXXXXXXX at xxxx|20|) command is executed and the destination number rings for 20 sec after which I receive as "503 Service Unavailable", but not "480 Temporarily unavailable". Dial(SIP/XXXXXXXXXX at xxxx|20|) exten => XXXXXX,n,NoOp(Dialstatus:${DIALSTATUS})

I have some problem to configure the call from asterisk to ser. [globals] SERADDRESS=xxx.xxx.xxx.xxx:5060 exten => 77,1,Dial(SIP/phonenumbertocall@${SERADDRESS},20,r) Error in Sip Debug ------------------------------------------- NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to '"Alexg"

Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two

Greetings. I have next feature in features.conf : send => *9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl What it does is parsing CALLERID and DNID from AGI input, performing some actions in MySQL with these values, and then running application for peer (for example, PlayBack) Sounds simple, and it really is. When my user is receiving a call (we are the B leg) and presses *9, everything

Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at

Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now !!!!! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions

Hi experts, I wish someone would kindly give me a hand on a problem on Asterisk Realtime. May I know how to enable the debug messages for the Asterisk SIP Registrar query the SIP user data in the created MySQL table. I found that I can see the debug message for cdr_mysql which shows it can connect to Mysql successfully, but can't find any for app_addon_sql_mysql. Million thanks in advance !

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote: > On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com> >> ha scritto: >> >> We experience the same thing. The solution we use is to not change >>>

Hi all, I`m searching for a special solution to send text messages inside Sip info packets, that are normally used for dtmf signalization. So far I?m able to exchange sip Info messages between two softphones which are connected directly together (only over a Switch). By connecting both Softphones on the asterisk pbx, registration is ok and the voice interconnection is also fine. During the call,

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

>Date: Sat, 11 Oct 2003 22:07:49 -0700 >To: asterisk-users@lists.digium.com >From: John Todd <jtodd@loligo.com> >Subject: RE: [Asterisk-Users] SIP / IAX over satellite > >[post re-ordered chronologically] > >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman

Hi all I am using Asterisk 12.4.0 on debian 7.6 x64 I experience some troubles with some specific calls, so I want to dig into this as deep as possible I run these CLI commands: > sip set history on (answer: SIP History Recording Enabled) > sip set debug peer 501 (answer: SIP Debugging Enabled for IP: ...) > sip set debug peer 502 (answer: same with another IP) but when I

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

Customer has a Verizon Business SIP trunk, I'm still used to PRI T1 myself for local service. The fax machines are having some issues (I can use analog phone to call out fine) and I'm checking on modem passthrough with Verizon, but wonder if any else is using Verizon Business for SIP trunk and what your faxing milage was? Did they support G711 and modem-passthough, etc? Also checking