similar to: Does anyone provide inbound UK numbers using IAX ?

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:

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

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

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 Community! If this issue was already topic, please excuse or delete my request... Topic 1 "no ringtone": I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

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 All, Can anyone recommend a supplier and/or a particular a model of SIP voip phones in the UK? I don't seem to be having much luck locating any. Thanks, Nathan.

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

Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the ?170 (exc vat) mark. Regards, Nathan.

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

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

Hi All, I am looking for a provider/s of inbound DID - IAX numbers, for UK, USA, and Australia. Preferably free or low cost :-) Can anyone make a good reference? Many thanks Chris PS: I appreciate this is perhaps a little OT, please feel free to reply off list. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, and Johannes if you're around, I was trying to set up the "new" antispam plugin with the mailtrain backend, and i was trying to use the sendmail binary that's part of a qmail system. The message wouldn't get sent and i kept getting exit error 100 from the /var/qmail/bin/sendmail binary. To try to figure out what was happening, i mean a fake sendmail program,

Running FreeBSD 8.3 -- and updating sources on a daily base and building everything -- I found a new APIC/ACPI problem introduced in the past week. I have a Toshiba Satellite U205 with an Intel Core Duo (not a Core 2). It used to work fine with both cores but then sometime in on the road to BSD 8.0 the machine began hanging. So I added to /boot/loader.conf hint.apic.0.disabled="1"

Hi, I'm trying to write a simple pass to print out Loop invariant instructions, using the Loop::isLoopInvariant(Instruction *I) function. it is giving me false value on instructions which should be loop invariant. the code i am using is: bool MyLoopPass::runOnLoop(Loop * L, LPPassManager &lpm){ BasicBlock* lat=L->getLoopLatch(); for (BasicBlock::iterator i = lat->begin(), e =

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]:

Hurray! Hopefully, he'll get put away for a while. Charged with SPAMming, money laundering, fraud, etc. Mike -- p="p=%c%s%c;main(){printf(p,34,p,34);}";main(){printf(p,34,p,34);} Oppose globalization and One World Governments like the UN. This message made from 100% recycled bits. You have found the bank of Larn. I can explain it for you, but I can't understand it for you. I