search for: progressinband

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.

...caller's phone until the remote phone is answered. I finally tracked this down to several things: a) My SIP provider sends "183 Session Progress" and inband ringback prior to sending a "180 Ringing". b) The default sip.conf file that ships with asterisk suggests using "progressinband=no" for polycom phones. c) The "progessinband=no" setting prevents the "180 Ringing" from being forwarded to the phone if it is received after the "183 Session Progress". d) Called-Parity-ID appears to be only sent to the phone with "180 Ringing" and &qu...

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortense...

...calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this suggests the call progress tones are being generated by the SIP device. As a result I have edited sip.conf to set "progressinband=never" but this has made no difference (even after a total restart). Previously I was running 1.0.7 without this problem, I upgraded to fix a problem with Monitor (the call stopped monitoring when transfered, 1.2.5 has fixed this). Does any one have any suggestions? - -- Ron Wellsted r...

Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg "30% increase") that would be great, rather than just "lots". Also, are there any ATAs which are known to not work with progressinband = yes? We have...

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 with 1.4.36/1.4.41...

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 allows early media through. That option has a complicated past. > > w/o the option progrssinband=never just the sip-pa...

...issing, if the second number (the second trunk) of the asterisk installation was used! The only difference between those two trunks is: The first trunk is configured to a ring group - the second trunk is configured directly to an extension. My solution after long time of digging around: I added progressinband=never to sip_general_additional.conf But this solution confuses me, because http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband tells: progressinband=never Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default...

....168.0.175 255.255.255.255 > 5060 > Unmonitored > > > Added to sip.conf: > > [175polycom] > type=friend > host=192.168.0.175 > defaultip=192.168.0.175 > dtmfmode=inband > mailbox=175 > context=sip > callerid="I am Don" > progressinband=no ;polycom's seem to have trouble with the > default progressinband=never > > [176polycom] > type=friend > host=192.168.0.176 > defaultip=192.168.0.176 > dtmfmode=inband > mailbox=176 > context=sip > callerid="I am a jerk" > progressinband=no ;polyc...

I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

...he phone. Instead I just see the usual RTP flows. I've been trying to fix this for hours, does anyone have any ideas how to get this working correctly? Asterisk version is 13.25.0 The settings I think are relevant (I'm using chan_sip): (sip.conf) ignoresdpversion=yes internal_timing=yes progressinband=never silencesuppression=no prematuremedia=no (Per peer) progressinband=yes directrtpsetup=no dtmfmode=rfc2833 directmedia=no silencesuppression=no prematuremedia=no TIA Mark. -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: &l...

...it's going to 192.168.200.99 (the phone). I've played with all sorts of settings in sip.conf, but the messages persist. Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid="HFT Booth 0 <(619) 364-4850>" allowsubscribe=yes And some of the Polycom phone config: reg reg.1.displayName="HFT0" reg.1.address="6193644850" reg.1.label="4850"...

...eo) that "their" SIP server is able to provide early video (using a Grandstream 3157v2 with "preview" enabled), but I would like to have this with asterisk... I'm currently using asterisk 12.2.x. I tried with all kinds of combinations of "prematuremedia" and "progressinband" in sip.conf and many different dialplan-extension-"scripts" but to no avail... sniffing with wireshark shows me, that the caller (doorstation) is sending H.264 video but the RTP video stream is not passed on to the callee by asterisk. (establishing a direct-video - without preview...

Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi

...to work properly? Any suggestions would be greatly appreciated. Here is the sip.conf file [test] type=friend secret=blahpoly insecure=yes host=dynamic qualify=500 nat=no mailbox=testmailbox callerid=Yourname <test> conext=local disallow=all allow=ulaw progressinband=no here is the local section of the dial plan. exten => 850,1,Goto(Mercury-Network,850,1) exten => 888,1,VoiceMailMain(@Mercury-Network-Emp)

I have Polycom ip-300 phones that worked yesterday but dont seem to work today (at least dtmf signalling once connected to the asterisk box) The current configuration is: [general] port = 5060 bindaddr = 0.0.0.0 context = test srvlookup = yes dtmf = inband allow = all dtmfmode=inband progressinband=no disallow=all allow=ulaw pedantic=no [202] type=user secret=xxxx context=test mailbox=202 host=dynamic [202] type=peer context=test secret=xxxx dtmfmode=rfc2833...

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 insecure=port dtmfmode=rfc2833 canre...

...don't appreciate your tone. Douglas. -----Original Message----- From: Andrew Joakimsen [mailto:joakimsen@gmail.com] Sent: Monday, December 11, 2006 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even withprogressinband=yes When we send 183, that means 'inband progress' is available. That does _not_ necessarily mean that it is ringing, it could be any sort of progress tone, or even audio from an IVR. If your ATA does not stop its own ringing generator and start forwarding the audio, it is broken. It is...

All, Using SIP with progressinband=yes I get Asterisk to generate the ringing sound for callers. However, I was wondering if it is possible to configure what is 'played back' to the calling party? i.e. instead of just 'ring ring' could I potentially play back a song from an MP3, WAV or GSM file? I'm thinki...

.... 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 asterisk systems are using 1.4. I have played >around with "progressinband" in sip.conf with now success. Whatever I set >progressinband to, it doesn't seem to change a thing. "183 Session >Progress" never seems to be called when looking at sip debug. It is only >when I use the Progress application before my dial command that I get the >&quo...