similar to: Dial timeout when server down

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

How frequent do the qualify packets go out? I've got 5 phones at a remote location with ADSL and their qualify is set to 5000. I've noticed that they will go "UNREACHABLE" and then immediatly go "REACHABLE" within a second or two. We have this theory that the ADSL link (SBC) "caches" its route to our network (where Asterisk is) and so we get bad ping

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

Two * servers: *a and *b. Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works. At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from? Thanks for any help in advanced, and sorry if

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

Hullo over there. i'm trying to link an asterisk box with a legacy PBX system with a four wire trunk line. the legacy PBX has 21 analog phones connected to it and i would like to route calls to another site via the asterisk box. i would like to use E&M signaling over this line. my question is how many simultaneous calls can you have over this line with E&M signaling. is there a better