similar to: Error Header field Via

Hi I wonder if anyone can throw some light on the * console message. This only occurs when I register a phone on the end of a BT ADSL line, with a Draytec router/modem. The phone registers okay but cannot dial out. Console message: Notice[1125329600]: File chan_sip.c, Line 1759 (copy_via_headers): No field 'Via' present to copy Thanks Steven *****************************************

Hi, I'm chasing a bug in chan_sip.c where Asterisk is removing the rport parameter out of the via headers. Here's my scenario: UA -> Snom NATf -> Snom 4S Proxy -> Asterisk Echo Test Function NATf, the proxy, and Asterisk are all on public IPs. So my question is: In chan_sip.c, copy_via_headers function, I see an if statement testing for "(ast_test_flag(p, SIP_NAT) ==

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hi, someone can help me, I am ot able to find the way how to user Ajax.updaterto test if the request give some positive or negative result. I am able only to return the result inside a div. An example is appreciated. _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me?

Hi, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm

There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/619b8f2b/attachment.htm

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

Hi, one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function. Any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.

Quick question for those who know the innards of chan_sip: Does chan_sip use the "To:" header of an incoming INVITE request, for anything other than setting SIP_HEADER(TO) ? As far as I can tell so far, the target extension is taken from the request URI, i.e. sip:extension at domain, and the target context is taken from the section in sip.conf that matches the request's source IP

Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, "sip:", 4)) { exten += 4; } else if (!strncasecmp(exten, "sips:", 5)) { exten += 5; } else { ast_log(LOG_WARNING, "Huh? Not an

Hi, I saw one site (bubbleshare) that it is able to caputer the click on the log in link, however, I cannot understand how they can do that Someone can explaint it to me? Thank you _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails-spinoffs

Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea?

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:pass@provider/400 [inside] exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT) exten => _4X.,2,hangup -- Executing

Hi. It is propbably a really naive problem, but I really couldn't find answer how to connect two Astrisks via SIP. I managed to do it via IAX without any problem. But this is a test installation and I would like to connect them via SIP. So I have two computers: pbx1 - 10.1.3.207 pbx2 - 10.1.3.204 pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to call user from pbx2 to

hello to asterisk start WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum retries exceeded on call xxxxxxx...xxxxxxxxxx@192.168.0.1 for seqno 102 (Request) with a call from x-lite 38113@fwd.pulver.com WARNING {98311] : File chan_sip.c, line 2002 (__transmit_response): Unable to determine sequence number from '' and x-lite hang up the second warning is new since morning