similar to: Same issue, different way to ask.

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64 very good, but since FC keeps updating, I tried to follow newer kernel versions. I can't pass the zaptel compilation. Everything is OK, but when I finished, and tried to load it, allways got module not found when I run modprobe zaptel, and modprobe ztdummy. I already tried to modify is with the sed 1 option but

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

This issue may sounds trivial I need to build a Router for send Internet + VoIP traffic. The computers are in a different network that the Phone Gateway. The Computers are going to be send to a 3 Mbps connection using OSPF, in the meantime the phones are going to be send to a T1 using OSPF too. The routing software is going to be Zebra. I need to switch the outgoing in case that the T1 or

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar

Hello Everyone, This is my first post, and this is my problem :-). I have a asterisk@home, work excellent (only internal users), but i need outbound calls. One person give me an access to his "Cisco 5300 Media Gateway", he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the

I need to find someone to work with me in the Grand Rapids Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get me contact info if you are interested. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/315eff3f/attachment.htm

This is all the reference to PTHREAD_MUTEX_RECURSIVE on lock.h #ifdef __APPLE__ /* Provide the Linux initializers for MacOS X */ #define PTHREAD_MUTEX_RECURSIVE_NP PTHREAD_MUTEX_RECURSIVE #define PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP { 0x4d555458, \ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0x20 } } #endif #ifdef BSD #ifdef __GNUC__ #define

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin

Before I go ahead and spend $40.000 on a Cisco 5400, just because my clients need T.38 faxing, I want to ask the community if there is any chance of having Asterisk receive G729+T.38 and sending the call via Zaptel to its final destination. Any answer will be appreciated. Federico

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the