similar to: Immediate 180 Response on Invite

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.

Chris Abel writes: >Hello everyone! > >I've had this problem for a while and cant figure it out. When an outside >caller calls an extension on my asterisk system, they do not hear any >sort of ringing. Inside extensions calling other extensions do hear >ringing. We have 3 other asterisk systems that are configured the same >way, but do not have this problem. We think it

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

Hello! The T.38 Digium Fax Driver sometimes responds with a successful sending of a fax, when in fact, the fax did not go through. 1. Where does this problem lie? 2. How to go about fixing it. Thanks, Elliot

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper

Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

Hmmm. Ok, that's true. At the very least it will create confusing CDR's I think... maybe. We're not billing our OnNet traffic at all. Only the traffic that goes OffNet, to our switch is billed (if it leaves our switch that is...). I was thinking earlier too that we only need progressinband on traffic that goes to the PSTN, via our switch. OnNet traffic will never generate reorder

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

Hello! I am having an odd problem in that when the caller dials extension "2" in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16

Hello, I have question regarding the changes that are made in the sip protocol in Asterisk - the option progressinband. When this option is set to yes in asterisk version 1.4.21.1 - the call flow is: sip.conf: progressinband=yes Device Asterisk -----------INVITE SDP---------> <---------100 Trying------------ <-----183 Session Prgoress-- After version 1.4.2X+ (tested

Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is " john at phone.com" needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Whoops, email client auto-filled dev previously. Let's try this again. Michael Maier wrote: <snip> > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: The default in 13 is "no" which still