similar to: SIP hangup on call proceeding message

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

Is there a way to allow a user to dial an extension after listening to your voicemail instead of leaving a message? Example would be the big boss is on vacation and changes his out message to say "you can reach my assistant at by dialing 1234 now or leave me a message". Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when AT&T is using CUSTOM? My understanding is that

Hi, here in Italy we have a special kind of phone services that are named "audiotel", those services are also known as "premium". Audiotel services are billed in a different way from normal phone calls and the revenue is shared between the answering company and the telco, they are used to give weather information, chat party lines and so on. Inbound lines are coming over

I am having an issue with the CallerID Number not being passed to my phone in the SIP packet. The CallerID Name is passed just fine and displayed on the phone with no issue. I have done a NoOp() in my extension.conf and successfully seen both the CallerID name and number correctly. So that leads me to believe that Asterisk is handeling it correctly. However, when I do a packet capture of the

Hi, The descriptions of Progress() and Proceeding() are really vague. Progress(): ---cut---------------- [Synopsis] Indicate progress [Description] Progress(): This application will request that in-band progress information be provided to the calling channel. ---cut---------------- Proceeding(): ---cut---------------- [Synopsis] Indicate proceeding [Description] Proceeding(): This

Hello R-users, In a loop, if a function, such as "nls", gives an error, is it possible to coerce R to continue proceeding the next command with the same loop? Thanks so much for your advice! Hanna Lu [[alternative HTML version deleted]]

I have a setup with Yealink phones & Asterisk Server (all latest patches). I am using BLF to display the states of other phones. While this works MOST of the time (busy, being called) it does NOT work when a phone is NOT regisstered at all, the yealink phones display a green dot EVEN if a phone is turned off (try explain this to users, they are shaking their heads!!!) I can see on the

Hi ! I have a TDM40B and i try to use it connected to modem for incoming call data transfert. I have no problem to use it with a phone and a talk communication work fine. But when we try to use with modem, with most modem, we got data carrier for few seconds and channel hungup. < [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1] -- Zap/4-1 is ringing << [ TYPE: Null Frame

I am sending SIP or H323 calls to a carrier, and I need to store in the CDR why the calls are rejected or why they hang up. In SIP, it can be code 503, 500, 488, etc. How do I get the information in my dialplan? I don't mean $(DIALSTATUS}, but the real numeric code F.Alves

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple extensions configured, but now I have to add one which uses the special h extension to perform a CURL

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? Thanks

7. Take an Asterisk training course and become a dCAP. As for "we have try to solve it but we're lack of asterisk knowledge" - would you get a plumber to service your car? Why not employ (as in 'pay money') somebody to investigate this further. As Satish pointed out - QoS type issues take a lot of debugging and that usually has to be done on-site. BTW - I doubt any of

Hey everyone, we are using several TE410 cards in a production environment that are connected to several operators PRI's and it works great.. For one of the operators we have seen some strange problems in cdr mismatches however. Our cdr's show phonecalls that are disconnected at a certain timelimit while the operators cdr's show the user has disconnected a lot earlier. I thought

One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for channel

Hi all, I have asterisk connected to my voice application server. Asterisk is connected and registering to a linksys spa400 box. I am running an application on a perticular extention (141). Here is a snip from my extensions.conf... exten => spa400,s,MyApp(/etc/asterisk/MyAppConfig.conf) exten => spa400,s+1,Hangup when an incoming call comes,It is accepted properly,And the application