similar to: Sip: problem authenticating (with Cisco VoIP IOS 12.x) [long]

Hi everybody, I'm trying to use Asterisk as SUA with SER. But Asterisk doesn't succeed to register the Asterisk user to SER... the standard error output of SER tell me: "ERROR: forward_msg: no 2nd via found in reply" the SIP message looks like: to 193.175.133.19:5060 Retransmitting #5 (no NAT): REGISTER sip:potemkin.fokus.fraunhofer.de SIP/2.0 Via: SIP/2.0/UDP

I have asterisk configured on a machine connected to Internet by a cable modem with a public ip. The same network card has a private lan address which I'm trying to use to play with an asterisk configuration with X-lite or an softip phone. sip.conf has bindaddr=192.168.0.1 [the private address of the LAN] to which the client [192.168.0.2] is connected. Doing a tcpdump on the register

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

I'm having problems getting Asterisk SIP to register with an Entice softswitch SIP Gateway. My provider tells me that all thats needed is a user name, password and the IP address and to register and it needs to be using MD5 authentication. I continualy get a "603 Decline" message. The provider of the gateway says they are not receiving any authentication information. Registration

Mark has fixed the REGISTER issues to be more RFC compliant. I've created a new thread so that those of you who got bored with the old thread might read this new one. The feature that has just been added was added a while ago, but now it actually seems to _work_. :-) If you have a SIP server to which you are trying to REGISTER, and they demand valid domain (the part after the

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.

Hello to everybody, Sorry if my english isn?t very good. I need pass 802.1q through a VPN between two offices. I have mounted a WRT54GL, with OpenWRT firmware, conected to a switch trunk port in both offices. In the switch of the first office I have created five tagged VLANs and I need pass these VLAN to the second offices where it has created it too. Can I do this with Tinc in HUB/Switch

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

I'm looking at RFC 3261, I think the latest SIP standard and have a few questions about the * sip implementation: 1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue a provisional response to non-INVITE requests.

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

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

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

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 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

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

Again rc, just in case I broke something. Or maybe I should just start creating nightly CVS snapshots. Anyway, fixes the few bugs people have reported lately: - Mails with nested MIME parts might have caused incorrect BODY and BODYSTRUCTURE fetches and sometimes might have crashed dovecot (assert at imap-bodystructure.c). If client had already successfully done the BODY fetching a

Several bugfixes since 0.99.6rc3. Here's again the change summary since 0.99.5: v0.99.6 2003-01-13 Timo Sirainen <tss at iki.fi> + THREAD=REFERENCES extension support. ORDEREDSUBJECT would be easy to add, but I think it's pretty useless. + SORT is much faster now. + mbox: If ~/mail directory isn't found, create it. + Log login usernames * Some coding style changes