similar to: 1.6 Troubleshooting help

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing

Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]:

Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission 8acaea6dc4c6e9b5 at

I'm struggle to answer a simple question. One user at extension 4000 wants a custom .gsm file to play for their mailbox. I can't figure where to put it/what to set in voicemail.conf to achieve this: voicemail.conf 4000 => 4000,system,voicemail at ....net Relevant extensions.conf line: exten => 2,n,VoiceMail(4000 at voicemail) It all works fine, playing the system VM greating, but

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting

Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

hi i got TOS and retranssmission error on receiving SIP call chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 for seqno 43156 (Critical Response) -- See doc/sip-retransmit.txt. [May 22 13:42:44] WARNING[18021]: chan_sip.c:2821 retrans_pkt: Hanging up call 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 - no reply to

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

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged

I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an

Hello - I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone =

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any