search for: sipgetheader

hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) Any one using this __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best...

I'm tring to use the function named sipgetheader in asterisk, but I downloaded the asterisk version 1.0.3 in which this function doesn't appear. What the simplier solution to my problem? May I download something else? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...

...Stopping retransmission on '9CB723DE-FD6111DB-9FBF9C45-B28951D9@128.91.56.38' of Response 101: Match Not Found [May 9 08:42:43] DEBUG[20530]: pbx.c:1795 pbx_extension_helper: Launching 'Set' [May 9 08:42:43] WARNING[20530]: pbx.c:1783 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 700, 4) [May 9 08:42:43] DEBUG[20530]: pbx.c:2393 __ast_pbx_run: Spawn extension (default,700,4) exited non-zero on 'SIP/128.91.56.38-09c6e8f0' [May 9 08:42:43] DEBUG[20530]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/128.91.56.38-09c6e8f0' [...

hello i m trying to use radius with asterisk http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth how to fix this patch 8. Make sure that your Asterisk includes all related bug fixes and patches, namely: - SIPGetHeaders for chan_sip (derived from chan_sip2 ) i m using asterisk-1.0.9 extensions.conf exten => _X.,1,SIPGetHeader(SIP_Authorization=Proxy-Authorization) ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football....

...#39;: Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' Problem 3) Sometimes the program crashes during (at end of) the startup sequence. Warning about "flexibel rate not heavily tested". Is this just a codec I can configure off/disable, or is this a crucial part of the system that will hopefully be fixed soon. I got this...

We've had a regular Park function in the past but recently I found the ParkAndAnnounce() application and I love the idea behind it. Here's a snip from the wiki (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) so that we're all talking the same language: || ParkAndAnnounce(announce:template|timeout|dial|return_context) || || Park a call into the

I need to understand which is the caller's identity. I'm tring to use the command named SIPGetHeader in asterisk, to get the >From Header of the Invite message, but I downloaded the asterisk version 1.0.3 in which this command doesn't appear. What's the easiest solution of my problem? May I download something else? Thanks in advance, Paolo -------------- next part -------------- An HTM...

Can anyone help me to solve the next problem???? I need to get the caller's identity from the Sip messages. I tried to use the command SIPGetHeader, but the downloaded version 1.0.3 doesn't work. Does a new asterisk version exist which solves my problem? I'm looking forward to having a prompt reply from you! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...

Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin.

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

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

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an external database (RADIUS - LDAP)", etc. Although, none of these seems to give me the

...09:09:15 NOTICE[6622]: chan_sip.c:12319 reload_config: Unable to load config sip.conf == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' == Registered custom function SIP_HEADER == Registered custom function SIPPEER == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action SIPpeers == Manager registered action SIPshowpeer [chan_agent.so] => (Agent Proxy C...

I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in new

...nel. As a result, a lot of the chan_sip2 code is now integrated in the CVS head. The next part to move into both chan_sip and chan_iax2 is the configuration templates. The latest release, called Yngve, has a few additions * SIPAddHeader(): An application to add a SIP header to an outbound call * SIPGetHeader(): An application to read any SIP header on incoming call These are really useful if you want to read RPID headers or transfer an accountcode between two Asterisk servers. I've also changed the authentication part and added realm based authentication. This way, you can configure Asterisk to a...

...149236666@67.149.187.174:5060 for 3600 -- SIP Seeding '2249' at 2249@24.145.226.232:15060 for 36 == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' [chan_modem_aopen.so] => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'A...

...149236666@67.149.187.174:5060 for 3600 -- SIP Seeding '2249' at 2249@24.145.226.232:15060 for 36 == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' [chan_modem_aopen.so] => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'A...

...el in Asterisk: * SIP realm authentication * Additional manager events * Improved DNS SRV support * SIP Digest Authentication based on digest user, not From: name * Templates for easier configuration of multiple devices * Mysql Authentication support * Support for Symmetric RTP * New applications: sipgetheader() and sipaddheader() The idea with the chan_sip2 project is to have a testing ground before new features are introduced in the ordinary chan_sip. A lot of features have been ported from chan_sip2 to chan_sip in CVS. In order to continue to do so, I need more feedback on the features I add. Add you...

...now produced two patches that adds support for adding and reading SIP headers in INVITEs to CVS head chan_sip. If this is useful for you, please test the patches and add a comment to the bug report. In order for this to be included in the CVS, it seems like I need some support from SIP users :-) * SIPgetheader: http://bugs.digium.com/bug_view_page.php?bug_id=0002838 * SIPaddheader: http://bugs.digium.com/bug_view_page.php?bug_id=0002846 *** SIP Patch: Outbound Proxy Support ------------------------------------- There are a lot of different SIP servers. Some of the most used are: * SIP registrar serve...

...': Found == SIP Listening on 192.168.0.8:5060 == Using TOS bits 0 == Parsing '/etc/asterisk/sip_notify.conf': Not found (No such file or directory) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' == Manager registered action SIPpeers == Manager registered action SIPshowpeer == Registered custom function SIP_HEADER [skipping chan_modem_aopen.so] [skipping chan_modem_bestdata.so] [skipping chan_agent.so] [skipping chan_mgcp.so] [chan_iax2.so] => (Inter Asterisk eXchange (Ve...