similar to: Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk

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

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

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

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]:

Since I so miserably misspelled packaging - a new thread specific to the issue at hand. I found a "workaround". In short, xlc does not accept arrays of nested array - the bottom line (best message) is: 1506-995 (S) An aggregate containing a flexible array member cannot be used as a member of a structure or as an array element. At the core - all the other messages come because this

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Hi, my dovecot installation works since months and clients authenticate using CRAM-MD5. But today I got the first chance to test an client that supports DIGEST-MD5 - and it doesn't work. Because of lack of other supporting clients and servers I'm now at the point I don't know which side is to blame. The error I get after the client answers the servers challenge is "-ERR

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

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

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

Calling all SIP gurus-- I'm trying to register my asterisk to an ISP's SIP gateway. I'm getting authentification errors. Here's the results of SIP DEBUG against it's IP. [I've tweaked all confidential fields so as to protect the innocent (namely, me).] --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name myfavoriteisp 12 headers, 0

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

George, I have the detailed log below. (Resending after trimming the email to 40KB.) The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? Thanks! --------------------- Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 Via: SIP/2.0/UDP 18.18.19.123:5060

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 list, We have a network with some XP and some Windows 7 computer, we use samba 3.6.6 on debian 6.0.6 from debian-backports. It's a pdc with passdb backend = ldapsam. In our logs there are lots of: ARCServer slapd[1263]: SASL [conn=46778] Failure: realm changed: authentication aborted I found out that at that time this emerges the tcpdump says: 12:59:54.656399 IP client.49551 >

Hello, When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to terminate the session is to send a BYE request to UA. After tracing the traffic on port 5060 UDP i recognized that my asterisk is NOT sending a BYE request to it's peer, so the peer doen't know to end the session and continues to send RTP packages to me. Does anyone know how to fix this? Here's

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. The registration works though, right? You might want to compare