similar to: asterisk console messages

Displaying 20 results from an estimated 1000 matches similar to: "asterisk console messages"

2004 May 22
1
Sip proxy registration help
Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded
2015 Apr 19
2
yum install failiure - CentOS-7 - Base
------------ Original Message ------------ > Date: Sunday, April 19, 2015 18:44:43 +0000 > From: Sarogahtyp <sarogahtyp at web.de> > To: centos at centos.org > Subject: [CentOS] yum install failiure - CentOS-7 - Base > > I have a running CentOS 6.5 64-bit system running and i like to > have a CentOS 7 chrooted system inside. > Ive done that chroot environment as
2004 Jan 13
6
SIP and AGI crash...
Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>
2019 Sep 05
1
install_github and survival
I treat CRAN as the main repository for survival, but I have also had a github (therneau/survival) version for a couple of years.? It has a vignette2 directory, for instance, that contains extra vignettes that either take too long to run or depend on other packages.? It also gets updated more often than CRAN (though those updates mght not be as well tested yet). In any case, since it is
2010 May 21
1
Hanging up call - no reply to our critical packet
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]:
2006 Jan 09
2
Question on Kernel boot options
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a project that uses syslinux to load the kernels from the CD to create disk and/or partition images. The process works great, but as I have been building newer kernel images with new disk and nic drivers, I have had users run into problems. Example: one users machine would fail at hotplug detection, creating a kernel with this feature off
2019 Sep 06
2
install_github and survival
I cloned therneau/survival and the installation failed since there is no definition for exported function survfit(). A file seems to be missing - there is survfit0() and survfit0.R but, compared to CRAN, no survfit.R. Georgi Boshnakov ---------------------------------------------------------------------- Message: 1 Date: Thu, 05 Sep 2019 12:53:11 -0500 From: "Therneau, Terry M.,
2013 Sep 19
2
The call is established but without exchanged voice packets
Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see
2009 Dec 24
2
1.6 Troubleshooting help
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
2004 Jan 24
4
retrans_pkt: Maximum retries exceeded on call
Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!
2004 Mar 16
6
Maximum retries exceeded on call
Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar
2017 Jan 28
4
Asterisk 13.13.1
On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >
2004 Mar 31
2
SER Asterisk problem
Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006
2004 Nov 30
2
Dual NAT for SIP
Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum
2003 Sep 24
10
SIP / GrandStream Configuration
Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The
2003 Dec 11
2
SIP retries
Is there a way to increase the number of retries or the time to help with this? WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103 (Request) WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103
2004 Oct 25
2
library gregmisc
I write to ask you an help about the package gregmisc. I saw the instructions, and I need some functionalities of this package, but I am unable ti download it. On friday I was able to download thte .zip, but R does not install this package, today there is no possibility to download it. What I have to do? Thanks Anna Maria Paganoni Anna
2007 Aug 06
1
sip issue with one way audio
I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason
2004 Jan 27
1
Cisco 7960 Problems
Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call
2009 Apr 13
2
retransmision error con asterisk 1.4.24.1
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