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Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -------------- next part -------------- An

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the

asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client at theirdomain@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server /

...ING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patched by elguero * Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP insta...

...ING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patched by elguero * Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP insta...

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

Hi all, i have had sip TLS with an own signed certificate (using the ast_tls_cert script) running on asterisk-1.8.8 - i then have updated to 1.8.9.3 - and now i get the message "FILE * open failed!" I have already recreated the certificates with the script - but still no luck... Does anyone here know the source of the problem ? best regards, Wolfgang Pichler

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

So far it happened two (2) times, last time yesterday. Opened case with Asterisk https://issues.asterisk.org/jira/browse/ASTERISK-24593 All FXO channels went off-hook, Operator FXS was constantly ringing. If I picked a handset there was a dead silence. If I put it back on hook it would start ringing. FreePBX interface would not operate properly. Specifically it was slow to authenticate, then

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)