search for: sipgetheaders

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 spam protection around http://mail.yahoo.com

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-users/attachments/20050113/761194e1/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,

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.f...

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

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

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...

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

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

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

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005

This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a community, spending extra time on finding bugs, solving issues, improving

Time to reboot and re-start Asterisk, well, hrrm, monthly, news. It's been a hectic fall with a lot to do, both before and after Astricon. At this time, we're preparing for two Astricon shows in 2005. And no, we haven't made a decision on where to run the European Astricon, not yet. I am preparing to travel to the USA again this coming week. Today, I'm spending my time finding

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give