similar to: Call problems

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip

Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password.

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

Hi All, I am getting this error when i try to compile the "Linphone" package by typing----- make. please help me i am feeling very frustrated with this error pasdt from 7 days i am getting this error. please help me. speexec.c: In function `speex_ec_process': speexec.c:112: `spx_int32_t' undeclared (first use in this function) speexec.c:112: (Each undeclared identifier is

Hello, I try to add some Entries via PHP to samba 4 AD LDAP. The insert work only party, some values like telephonenumber, ipPhone and facsimileTelephoneNumber are not set. ldap_add always return success. Is there a way to see whats going on in ldap and whats wrong? I have try to set ldap_set_option($connect, LDAP_OPT_DEBUG_LEVEL, 7); in php, but it doesn't output more infos. Best

Hello This might not be a 'pure' * question, but it is relevant to general VOIP technology. I tried x-lite on my notebook with wireless connection(802.11). The software has been tested with the fixed line connection. It worked fine to call through *. When using wireless connection, it is clear on my side using notebook; however, there is loud noise on the other side of the call which uses

Dear all, I have PBX with asterisk 13.x a couple of IPPhone that connect to that asterisk PBX send an alphanumeric dialed phone number. for example, in my CDR table, field DST, it show dialed phone number like - 0C81318304632C (it should be 081318304632) - 08D11157112 (it should be 0811157112). Why it's happening ? and how can I prevent it to happen ? Thanks in advance, Ikka Jakarta

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

I wanna connect proxy server. my IP Phone -> my asterisk -> service provider's proxy server -> extern PSTN phone but asterisk server can't register to proxy server. I think that configuration is right. When asterisk send to register request, proxy server don't response. I did capture packet. but no response. MY setting sip.conf [kms]

Hi to all, I've just received my Cisco 7905G ipphone. I want to connect it to asterisk server but it looks that it has been preloaded with sccp protocol, so I guess I need H.323 or SIP firmware image of some kind. I have a working tftp server on my asterisk box also....What do I need to do now to get things wokring? Thanx in advance, Victor...