similar to: no ringtone - just silence/bridging of external calls

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission SIPCALL-435578583-1984100284 at 72.20.112.114 for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]:

Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support

Hi All, Is anyone use the sipcall.co.uk 'professional' account with a UK geographic number? What do you think of the service? Alternatively, who else are you using to terminate a UK geographic number on asterisk? Thanks, Nathan. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.529 / Virus Database: 324 - Release Date:

I have SIP registrations working correctly for FWD and Sipphone, but it is impossible to connect to Sipcall or Nikotel, I saw that someone on the list has problems with ICH. To try and sort out the problem I tried to register to Sipcall with Linphone and sent the dialogs to tech support of the equipment provider. Here is their answer:- The reason the registration fails is because not

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

i could add "r" option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/77a4d48f/attachment.htm>

Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client ------------> Asterisk ---------------> OpenSIPS So Asterisk keep a list of registered clients and only allows them to

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hello, I have uploaded a custom ringtone to our SPA-922's for the purpose of sounding like a door bell chime when the doorbell is pressed. I am using __alert_info to set this ringtone. It appears that I can only set the ringtone via alert_info for the ringtones that come from Linksys. Has anyone else seen this issue? I am doing the following: exten => 100,1,SetVar(_ALERT_INFO=doorbell)

We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone.

Hi All, Is there anyone providing UK geographic numbers that can be terminated on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or 02, not 08xx). I've tried the sipcall.co.uk service and it looks very good when using X-Lite but it will not work with Asterisk. Switching to IAX should also resolve issues around NAT - hurray! -Nathan

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue,,,,20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

Hello, would like to have distintive ringtone for internal calls, google gave me blurr answer. My extensions are 46**, any calls made within 46** I want to ring differently than external calls. Thanks in advance.

Sorry to bug the entire list with this as this is really a question for those who have been sucessful in configuring [*] to place and receive a SIPCALL call. Everying looks right in my config, I can see it registered etc but when I try to place the call I get: -- Executing Dial("SIP/100-2371", "SIP/8703409095@sipcall/04") in new stack Apr 29 22:50:34 WARNING[27089840]:

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten