similar to: asterisk-users Digest, Vol 54, Issue 107

Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

I have four-interface Shorewall config set up. The "dmz" interface is bridged with "net" so I can assign public IP''s to the servers in the DMZ. I opted to do this rather than SNAT or ARP proxying because one of the servers runs Asterisk and SIP and NAT don''t always work well together. Somehow, my firewall config is causing a one-way audio problem in

Hello, Let me first start by saying Shorewall is awesome, and I use it everywhere from single box firewall, to home network firewall, even to our corporate firewall. I am experiencing a problem getting my home firewall to work with my BroadVoice VoIP connection. I use the Sipura SPA-2100 ATA (Analog Telephone Adapter) that came with my BroadVoice account. This happened when I tried to replace

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=503 siqhamo@newlunar.co.za changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are

hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=532 kaber@trash.net changed: What |Removed |Added ---------------------------------------------------------------------------- AssignedTo|laforge@netfilter.org |kaber@trash.net ------- Additional Comments From kaber@trash.net 2007-01-26 19:45 MET ------- (In reply to comment #0) >

I have a working LVS-Setup on CentOS 5.4 with the following settings in sysctl.conf: net.ipv4.conf.lo.arp_ignore = 1 net.ipv4.conf.lo.arp_announce = 2 net.ipv4.conf.all.arp_ignore = 1 net.ipv4.conf.all.arp_announce = 2 Now I''d like to use shorewall, but after activating it, shorewall changes both arp_ignore values to 0. I just found out how to set arp_ignore for separate interfaces, but

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=532 Summary: ip_nat_sip rewrote Call-ID instead of Contact - patch attached Product: netfilter/iptables Version: linux-2.6.x Platform: All URL: http://ibp.de/ OS/Version: All Status: NEW Severity: normal Priority: P2

Hi. there is a way to MARK udp VOIP (SIP) traffic, in order to put in a highest prio class ? Traffic flow seems start on udp 5060 port, but next both server and client seems jump to a random(?) port. I can''t use CONNMARK because is udp traffic. I only see a pattern for L7 patch in order to SIP traffic identification , but I run 2.4 kernel series . When you patch 2.4 kernel with

Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Hi, while using cups as a print system and sharing all the printers with samba to windows clients, is it possible to 'view' a newly created printer in the browse list without restarting smbd ? (samba 2.2.7) Holger

I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to

Hey guys, I'm looking for some realworld specs on somebodys machine that will work with the Digium 4-port T1/PRI card and that will support 96 simultaneous calls. Dell is soon to release the PowerEdge 1850: 2U, Dual 3.6Ghz Xenon, 1Gb DDR2 RAM, Dual 36GB Ultra320 SCSI RAID, Hot swap Powersupply, one 64bit 133Mhz PCI and one 64bit 100Mhz PCI for about $3,000. Tack on a 4 port Digium card and

Hi all, I have a question concerning ActionWebService XMLRPC servers: Is it possible to send multicall requests to the Web service? I tried to use multicall and get the error message: no such method ''system.multicall'' on API [MyAPI] In Changeset 2021 there is the following commit message: add ''system.multicall'' support to XML-RPC. boxcarred methods must

Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service,

I set up dovecot on two servers with replication enabled. The replication is over a vpn connection. I get massive segfaults from doveadm-server every few minutes. dmesg: doveadm-server[30030]: segfault at 0 ip 0000557fa16d2e62 sp 00007ffdcaafec50 error 4 in doveadm-server[557fa16a3000+41000] [486397.225636] Code: 40 03 4c 8d 3c c3 48 8d 44 24 18 48 89 44 24 08 0f 1f 84 00 00 00 00 00 49 8b bc 24

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the