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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

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

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > <snip> > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee - but the callee doesn't send anything (not >> surprising, because there has been >>

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 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

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

Hello, My company is doing some * testing with our Class 5 softswitch and had some questions regarding ringback being provided to our PSTN users (off --> on net calling) Currently with MGCP subscribers, we know the PSTN ringing is provided by a digital PBX for example, However, it looks like with SIP, our softswitch is relying on MGCP signaling on our PSTN gateways to provide ringback

Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

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:

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.

For the past week or two, our customers who have Polycom phones have been experiencing a problem... but our customers with Cisco phones do not have this problem. The phones in question are: Polycom SoundPoint IP300 (firmware 1.3.1 or 1.3.4) Polycom SoundPoint IP500 (firmware 1.3.1 or 1.3.4) Cisco 7960 (firmware 7.2 or 7.3) The problem is this: when our Polycom users dial _some_ PSTN numbers,

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.

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

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

Hi All, I've been looking for information on how to use asterisk and early media to allow for a video-preview of the caller at the callee's phone for days... but I haven't been too successful :( I found that there seems to be a company "2N Helios IP" which claims (youtube-video) that "their" SIP server is able to provide early video (using a Grandstream 3157v2

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

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

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of