similar to: Help - create_addr_from_peer: 'UDP' is not a valid transport for 'exten1'. we only use 'TLS'! ending call.

While working on SPTPS UDP relaying I realized that there is one issue I didn't account for, which is that the sending node only knows the PMTU to the first relay node. It doesn't know the PMTU of the entire relay path beyond the first hop, because the relay nodes don't provide their own PMTU information over the metaprotocol. Now, in the legacy protocol this is not really an issue,

Hi Guys, i have a problem with PXE/UEFI Boot. Legacy PXE Boot is working fine. I use Dnsmasq with integrated tftp. As Mentioned above, legacy pxe boot is working fine, tcpdump shows: --- start tcpdump legacy pxe boot --- 13:44:16.589197 IP 0.0.0.0.bootpc > 255.255.255.255.bootps: BOOTP/DHCP, Request from f8:ca:b8:06:7d:ed (oui Unknown), length 548 13:44:20.543910 IP 0.0.0.0.bootpc >

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

Hey all, I've recently updated an Asterisk installation to 11.10.2 (from the Gentoo packaged build), and the update has (seemingly) broken TLS transport for snom phones (I could reproduce on snom 3x0 and 7x0 with 8.4.3x and 8.7.x.y firmwares). What happens is that the registration to the PBX is successful, and inbound calls work normally (i.e., when the phone is being signalled that a

I have used SSHD from openssh-3.7.1p1 on HP-UX 11:11. It works correctly and the entry in the logfile is: Sep 24 07:01:20 garm sshd[6625]: Accepted password for japs from 192.38.97.131 port 2463 Next I have upgraded to openssh-3.7.1p2 and restarted SSHD. It does not accept the password any more and the entries in the logfile are: Sep 24 12:21:38 garm sshd[19542]: User japs not allowed because

I don't think it's the snom, (the break key is set to "off") the "#" key is not being interpereted by the PBX as an attempt to initiate a transfer. Is this an error in my extensions.conf? Brian > >Message: 4 >Date: Wed, 15 Dec 2004 19:39:39 -0500 >From: Info <info@idatasys.com> >Subject: Re: [Asterisk-Users] Help with transferring a second call

Hi, I'm using Asterisk with two Cisco 7960 phones using SIP. I'm seeing the following weird behavior: SIP Phome 1 is extension 4002 SIP Phone 2 is extension 4003 I call 4002 from 4003 and that works fine. I call 4003 from 4002, and it rings locally to 4002, never gets to 4003. I'm able to send a config query packet to 4003 from the asterisk console and get a response, when I send

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

Hi all, Which ports do I need to have open on an NFS client's firewall to allow it to connect to a remote NFS servers? When I disable iptables (using ConfigServerFirewall), it connects fine, but as soon as I enable it, NFS gives me this error: root at saturn:[~]$ mount master1.mydomain.co.za:/saturn /bck mount: mount to NFS server 'master1.mydomain.co.za' failed: RPC Error: Unable to

I'll test it. I'm a bit confused with cgi-bin. I have made a lot of hosts.conf copies and didn't understand where does upsstats trying to find hosts.conf # ps -aux | grep ups _ups 13934 0.0 0.0 484 920 ?? Ss 5:41PM 0:01.53 /usr/local/bin/riello_ser -a senpro _ups 9171 0.0 0.1 532 1168 ?? Ss 5:41PM 0:00.13 /usr/local/sbin/upsd _ups 16906 0.0 0.1

Good afternoon, I am trying to communicate with a new Tripp-Lite UPS via usb and having little luck. After more than a week on this I've decided to ask for help. I am very inexperienced at posting information to message boards, so hopefully I'm doing this right. I assume this is the correct place to post this message. I will start with the equipment I have. I am using Debian Sarge

Hello All, I was wondering if somebody could elaborate the change in translation of codecs specifically the amount of time increased in Asterisk 11. For example *Asterisk 11* * **alaw **speex * *gsm **15000 **15000 * *ulaw 9150 15000* * * *Asterisk 1.6.x* * **alaw **speex * *gsm **2 12002 * *ulaw 1 12002* I did recalculate the

Hi, Hi, I'm having trouble transferring FSMO roles "DOMAINDNS" and FORESTDNS with below showing: samba-tool fsmo transfer --role=domaindns ERROR: Failed to delete role 'domaindns': LDAP error 50 LDAP_INSUFFICIENT_ACCESS_RIGHTS - <00002098: SecErr: DSID-031523E0, problem 4003 (INSUFF_ACCESS_RIGHTS), data 0 > <> root at samba4-dc:~# samba-tool fsmo transfer

Hello, A little informal input on the TCP/UDP checksum deferral/offload: it fails for me. I have one Xen host, running a snapshot of xen-unstable at 2006-03-24 (9435:11fee62328cc). Each of its domUs are allocated their own /30, with the help of a hacked-up vif script. Prior to working around the problem, Domain-0 could establish TCP connections to other domains. In the case of SSH, it would

Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data

Hi Gang I gave up on running asterisk with two interfaces without it mixing up the ip addresses. So I have removed one transport definition from pjsip.conf Now * keeps complaining: res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-name' I did a grep on /etc/asterisk for that transport name. It's in any file anymore.

Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From:

https://bugzilla.samba.org/show_bug.cgi?id=6508 Summary: Inconsistent exit code behavior between local, ssh transport, and rsyncd transport Product: rsync Version: 3.0.4 Platform: All OS/Version: All Status: NEW Severity: major Priority: P3 Component: core AssignedTo:

All, A bit of background: I work on a QA API on a network that is very choppy (a lot of network interrupts), and we use ssh to do a large part of this automation. This leads to some problems: ssh connections seem to be sensitive to network state, becoming unusable if the choppiness reaches a certain threshold, and either timing out or disconnecting if this happens. Anyways, I stumbled across