search for: sipcall

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

...des [general] allowguest=no alwaysauthreject=yes sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes,redundancy,maxdatagram=400 directrtpsetup=yes disallow=all allow=ulaw allow=alaw and the corresponding Peer [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 the Dialplan [inbound] exten => _X.,1,Answer() exten => _X.,n,Set(DB(lastcaller/numb...

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...

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.gris...

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]: chan_sip.c:901 create_addr: No such host: sipcall/04 Apr...

I finally got sipgate.de working for outgoing calls. I can dial and talk to any destination supported by my account but I can't hear the other party. My asterisk is behind a firewall router with forwarded rtp ports 10000-20000 (which is also defined in rtp.conf) to the * server. Incoming calls work fine and on clients in the same network directly registered with sipgate.de there are no

...00.3> Call-ID: 797316263@192.168.100.13 Content-Length: 0 User-Agent: kphone/4.0 Event: registration Allow-Events: presence Contact: "myusername" <sip:myusername@192.168.100.13;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" SipCall: Incoming request SipCall: New transaction created SipTransaction: Incoming Request SipTransaction: Retransmit 1 (4000) SipClient: Sending: 21:47:49.456 -------------------------------- REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: "sjphone2&...

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 problem than me and a resolution for it? --------- Topic 2 "external bridg...

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 all of t...

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]: chan_sip.c:1972 retrans_pkt: Hanging up call SIPCALL-435578583-1984100284 at 72.20.112.114 - no reply to our critical packet. == Spawn extension (ServiceNumbers, 104, 7) exited non-zero on...

...ng 'demo-congrats' (language 'en')", nothing else. On the other hand, kphone finishes their log with that: ===================================================================== ... res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 127.0.0.1:13998 ERROR: Open Failed ** audioIn: openDevice Failed. CallAudio: Creating OSS->RTP Diverter dtmfsenderTimeout DspAudio: Broken pipe (b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(...

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 draw this dialpan in the extensions.conf Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1 OK 0...

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

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

...an "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) > 8. Re: Caller entered digits ignored during >...

...an "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) > 8. Re: Caller entered digits ignored during >...

...FC35 fwd-outgoing xxxxxxxx from-sip No No -- Registered SIP 'jackie.clough' at 192.168.1.10 port 5060 expires 180 In extensions.conf the context from_SIP_extensions could be something like:- [from_SIP_extension] include => external include => sipcalls include => SIPextensions include => voicemail and the context external would be:- [external] ; Dial 9 for an outside line. Allow any call for now exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) exten => _9.,2,Playback(invalid) exten => _9.,3,Hangup Where g1 is the group 1 I defined in zap...

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone