similar to: Packet Testing

Displaying 20 results from an estimated 20000 matches similar to: "Packet Testing"

2006 Apr 05
1
IAX2 Origination Problem
Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when
2006 Jun 09
3
Compiling SVN Trunk
I have the same problem on some modules. For example app_math.so [app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728 __load_resource: missing mod_data for app_math.so Any help?. I have been looking , but nothing reasonable found. Thanks -- Alberto Sagredo
2006 Apr 04
1
IAX connection refused between 2 asterisks 1.2.5
Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta
2006 Apr 04
2
Can't get Pickup app working
I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",
2006 Apr 10
1
"chan_iax2.c: Ooh, voice format changed to ..."
Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723 to alaw DEBUG[15015]
2006 Apr 10
2
Wanted any /all used out of service Digium boards Mark
Wanted any /all used out of service Digium boards Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060410/3892463b/attachment.htm
2006 Apr 11
2
G726-40 required - Help!
Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager already offered this to the customer and now i do not know how to do it... Thanks a lot in advance,
2006 Apr 14
2
How to get 1.2.7 asterisk
Hi, Does "cvs checkout asterisk" gets the later version of asterisk? I tried "cvs checkout -r v1-2-7 asterisk", and didn't work for me. The only thing works is "cvs checkout -r v1-2 asterisk". What exactly is version tag for version 1.2.7? Thnx
2006 Apr 24
1
Change name User-Agent
Somebody knock where i change name?s User-Agent Thank?s -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 2703 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060424/3684c52f/attachment.bin
2006 Apr 18
2
eyeBeam + ASterisk 1.2.7.1 + Instant Message
Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: "User not available". In asterisk console appears a message saying: ------ Apr 18 17:13:22
2006 Jun 05
9
IAX Passing Variables
Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want
2006 Apr 28
1
Integrics release Enswitch 2.0
Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it
2006 Apr 29
2
How many asterisk process's are "normal"?
Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a "production" machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip):
2006 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File
2006 May 29
2
Problem with IAX2 dialin with portunity
Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>
2006 Jun 12
5
IAX DID channels as incoming hunt group?
Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the
2006 Apr 20
3
Asterisk Won't start after SVN Trunk Update
Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it
2012 Nov 16
1
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603 at DLPN_AlDimnaDialPlan:601] Dial("SIP/601-00000002", "SIP/603") in new stack [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433 dial_exec_full: Unable to
2015 Oct 11
2
same sip username with realms and chan_sip
Ludovic Gasc wrote: > Hello, > > same sip username with realms is possible with Asterisk ? > I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and > now, Asterisk crashes. Did PJSIP crash in general (it's usually a build problem if that happens) or was it when you were experimenting with different realms and such? -- Joshua Colp Digium, Inc. | Senior
2023 Mar 01
2
RTP address learning and timing problem
On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and