similar to: no nat, but one way only audio

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other

Hi all, I just configured a quescom 400 to route all gsm incoming calls to asterisk, now i would route all outgoing asterisk calls to gsm port of the quescom. Anyone has any idea how implement it? I did a configuration but i always get this error -- Got SIP response 503 "Service Unavailable" back from <ip_add_quescom400> Thanks in advance. Giordano

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si,

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

Hi there, I'd like to know if some of you knew Quescom products (http://www.quescom.com) and if it is possible to make run Asterisk with a Quescom400. Thanks. -- Asega Adrole

I terminate some calls on a h323 device (a quescom gsmgateway) from asterisk 1.2.3 with ooh323, the customer is complayining about choppy sound on most of the calls, the only warning message I can see is : src/chan_h323.c:944 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_102 (the calls sounds perfectly with iax/zap termination and the quescom seems to work fine with

I've ereditated this quite messy openldap server from the previous administrator, samba (3) relies on it for acting as a PDC. The main problem (while I build a new directory from scratch) is you can't add a machine account to the domain : On the client it says the credentials are invalid, anyway the real problem (from samba logs) seems to be : "Got too many (2) domain info entries

Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone