similar to: Sending bye to not establishment session

Hello, We are using this Asterisk: 1.2.14-BRIstuffed-0.3.0-PRE-1y Everything works fine but we have an issue (not often, but one call every some hundreds) I sniffed the communication between phone, Asterisk and softswitch. I can see that Asterisk receives a Cancel from phone but Asterisk never sends a Cancel to Softswitch. This makes us some problems: billing system doesn't allow next call

I am trying to use asterisk for an shared line gateway. When moving from one phone by placing the call on hold then having a second phone pickup that held call by sending asterisk a replaces header (http://www.ietf.org/rfc/rfc3891.txt) Asterisk does not seem to send a "bye" message to the original UA leaving the first phone stuck in a holding state. Am I missing something here? Here

Dear All I have some doubt about socket establishment. I am sending this question again. Sorry to bothering you a lot. Example : make.socket(host = "localhost", port=9754, fail = TRUE, server = FALSE) *Error in make.socket(host = "localhost", port = 9754, fail = TRUE, server = FALSE) : socket not established *Can anyone please help me to solve this error. Any help would be

Amongst others the asterisk server is connected via a BRI channel (EuroISDN) to the carrier. It takes a very (noticeable) long time until an incoming call is established and the phones being connected to Asterisk start ringing. As a test scenario I directly connected two different native BRI (ISDN) phones to the delivery point (FXS port) of our carrier. I started a call from my cell phone and as

Hello, I'm currently on Dovecot 1.1.17 and I have a user that runs into a problem quite a lot. She's using Thunderbird on a Mac. When she sends mail, Thunderbird is set up to save sent mail to her Sent folder. Often (usually when she has not sent another mail for a long while) when she sends mail, Thunderbird sits there saying "Copying message to Sent folder" for about thirty

Hello. I tryto use Statis at my dialplan to run my app (a) When Statis is running from making call ( I dial from softphone some exten and run dialplan context where call Statis(MyApp)) Asterisk responsed: ERROR[61517][C-00000019]: res_stasis.c:852 stasis_app_exec: Stasis app 'MyApp' not registered How I must Register MyApp -------------- next part -------------- An HTML attachment was

Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

Endpoint sends an INVITE Asterisk send an INVITE to the Carrier Carrier is down, does not even sends ACK PJSIP sends several INVITES End point sends <--- Received SIP request (397 bytes) from UDP XXXX::50187 ---> CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0 Via: SIP/2.0/UDP xxxxxxx :50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport Max-Forwards: 70 To: <sip:xxxxx at xxxxx> From:

On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > Hello - > > I am trying to decide if I have stumbled across a bug in PJSIP or I am > just missing something. My Asterisk has two interfaces, an "internal" eth0 > and an "external" eth1. In pjsip.conf, I define the following transports: > > [trusted] > type=transport >

Hello everyone, While using the following set of commands, I am having the error as below: FETCH 7 (X-GM-MSGID) A15 FETCH 7 (X-GM-MSGID) A15 BAD Error in IMAP command FETCH: Unknown parameter X-GM-MSGID Can I somehow disable such errors so that Dovecot won't send BYE on X-GM-MSGID but just proceed with following emails? -- My configuration is below: [root at server ~]# cat /etc/issue

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton <rnewton at digium.com> wrote: > On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > >> Hello - >> >> I am trying to decide if I have stumbled across a bug in PJSIP or I am >> just missing something. My Asterisk has two interfaces, an "internal" eth0 >> and an

Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an "internal" eth0 and an "external" eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use

Dear all, I have a question about session timers. I have one of my installations (* 1.6.2.7) where all SIP calls get stuck, like this: cs4wall*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry 192.168.40.178 42 3c291b87c66e-sl 0x8 (alaw) No Rx: BYE 192.168.40.179 41

Hi All, We have a Cisco CME linked to our Asterisk PBX (named 'epstein'). Off said PBX we have numerous other PBX's, some IAX and some SIP. On a call placed from CME (SIP) to 'epstein' it all works fine except for a few quirks. When calling through epstein to an IAX peer we get '100 trying' followed by '180 ringing' sent back down the SIP leg to CME.

Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. cheers - Ben.

Greetings When I start firefox or mozilla I get an hourglass for 3 seconds then nothing happends. Dropping to a terminal and typing firefox just returns to the command line. I re-installed firrefox and it installs fine until I quit then the same situation happends. After re-installing firefox it will run but the same situation happens where it won't start. I have installed int my home