similar to: Bad behaviour between X-Lite 3.0 and Asterisk

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I've been using for a long time asterisk-perl-0.08 for prepaid card applications, and I've identified a problem with the last releases of asterisk-1.2, installed with Trixbox. The command get_variable() raises a signal SIGPIPE when it is called (whatever the variable to get). I made tests with Asterisk 1.2.20, 1.2.21 and 1.2.22, and I

Hello, I use .call files in /var/spool/asterisk/outgoing to initiate calls automatically. And I'd like to setup the language used for the call in this file but I haven't found any way of doing this. I tried something like "Set: language=fr", "Set: ${LANGUAGE}=fr", ... but nothing worked. Is that possible? -- Beno?t M?rouze Ing?nieur D?v?loppement

Hi, I'm using the Perl AGI interface for a prepaid card platform. And sometimes (almost twice an hour), asterisk doesn't detect a call has been hung up. The call is so hung up when the time limit for the call is reached (the corresponding prepaid card is then emptied ...). I've tried to look in the asterisk log files to find anything suspect with these calls, and I've found a

Hi, which OSX softphone do you use that supports IAX2 protocol with Asterisk? thanks Mike

I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me

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]:

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these

when I make a call to the pstn it shows me this error: aximum retries exceeded on transmission 9d4a24f8-b673756b at 192.168.10.19 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 11 20:35:34] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging up call 9d4a24f8-b673756b at 192.168.10.19 - no reply to our critical packet (see doc/sip-retransmit.txt). bug? voicemail same

I'm watching my backup via rsync, throttled to a very low speed. Looks like downloading apache logs is taking the longest time (when I'm rsyncing over an old copy of the same data) because it's not noticing that, for example, www.chaosreigns.com-access.log.196.gz on the origin is the same file as www.chaosreigns.com-access.log.186.gz on the destination, so *everything* is getting

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

What is the problem with SIP retransmits? ----------------------------------------- Sometimes you get messages in the console like these: - "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet." - "retrans_pkt: Cancelling retransmit of OPTIONs" The SIP protocol is based on requests and replies. Both sides send requests and wait for replies.

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

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

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