similar to: Why are sip-channels too lagged?

Hi, my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this "normal"?: -------------------------------------------------- zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353, wanted 8 got 7), probably a buffer overrun. zaphfc: dropped audio (z1=2712, z2=2695, wanted 8 got 17, dropped 9). With Asterisk running (asterisk -vvvvvgc)

Hi, is anybody running asterisk on a xen domU and can give an opinion on the following: I have delegated a FritzCard and a HFC card to my domU and installed an asterisk setup that was running on the same isdn hardware but on a dedicated machine flawlessly. I experienced what I believed to be timing problems: sometimes calls on the Fritzcard did not seem to "reach" asterisk, when

Hi, can anybody comment on the cost of matching with IPP2P vs. Layer7. Also, does a iptables rule with more complicated matching mechanism also slow down processing if all the packets are matched before they reach the rule. I.e. is the mere existence of a potentially costly rule already slowing down processing or only if packets are actually processed by it? Thanks very much in advance.

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

Hello, can anybody interpret what the following means: [root@funke ipp2p-0.8.0]# iptables -t mangle -A MarkList0x666-ipp2p -p tcp -m ipp2p --edk -j MarkSet0x666 iptables: Unknown error 4294967295 ----- I have installed ipp2p-0.8.0 via: make copied ipt_ipp2p.ko to my kernel lib dir copied libipt_ipp2p.so to my iptables lib dir insmod ipt_ipp2p gives the following in dmesg: IPP2P v0.8.0

Hi, I defined several xen domains with libvirt. Can anybody tell me where libvirt stores these definitions? E.g. what files would one need to back up? Is it correct, that the xen domain definitions in /var/lib/xend/domains are overwritten regularly by libvirt? Many thanks, Arik

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

Hi, I have a Fedora 7 xen system with a domU to which I have delegated via pciback two isdn pci cards. One is a HFC-S card, the other a AVM Fritz!Card Pci 2.0. In the domU I load mISDN drivers for the cards. The two cards are recognized and function well UNTIL I either decide to unload one or both of the card driver modules (hfcpci and avmfritz) or I decide to shut down the domU. At this

Hi, I am having problems creating VMs in virt-manager. I have played with different xen hypervisor versions 3.0 & 3.4 (with rpm -i --force) but have in principle gone back to the originals. - Though I cannot absolutely exclude that something has gone wrong. In principle I am using rhel 5.4 with the gitco xen 3.4 hypervisor. Can anybody put me on the right track with the attached error

Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT

Hi, I was successfully using VoIP Buster via IAX2 for several weeks now. Yesterday/today it spontaneously stopped working. Using the "real" client the connection works well though. Anybody else experiencing this problem? Or asked differently: Is there anybody for whom it is still working? Can anybody tell me what the problem could be from this: -- Executing

Hi, I am having problems establishing an adsl connection in dom0 (but ONLY in dom0, without xen everything is ok). Following is my setup/situation: 1. Installed clean Fedora 7 (with virtualization option selected) 2. Booted with the non-xen kernel & setup adsl. Adsl connection works without any problems. 3. Rebooted with xen kernel into dom0. 4. adsl connection can no longer be activated

Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine

I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the

Hi, I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? Thanks for the help, Arik

Hi, following I would like to implement: 1. I receive a call. 2. I hang up the call. 3. I execute a macro I thought about using call files first... but they don't support macros, or? Then I figured I could use php agi after I receive the call, hang up the call with php agi and execute the macro with "exec ...". Unfortunately, the php agi seems to die with the hangup. Does

I have an asterisk server and I am trying to set the server up as a redirect server of all my internet SIP phones. My Asterisk server as his own internet IP address. At this moment I can make international calls to a IAX provider but I am now trying to setup a SIP provider as well And I get the following error -- Executing Dial("SIP/t10002-4666",

Hi, I am looking for inexpensive isdn card that supports nt mode with asterisk. Does it mean that if mISDN has nt mode support for a specific card that asterisk has too? If yes, I presume I could buy any of the cards listed on the PBX4Linux page: http://isdn.jolly.de/cards.html Or will asterisk work ONLY with cards with hfc chipsets? Cheers, Arik

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the