similar to: CID

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2)

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue?

I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

The latest libpri is supposed to handle this properly, but doesn't seem to. Here's the debug info. CALLERID(name) is set to empty. < Protocol Discriminator: Q.931 (8) len=66 < TEI=0 Call Ref: len= 2 (reference 256/0x100) (Sent from originator) < Message Type: SETUP (5) < [04 03 80 90 a2] < Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability:

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried

libswfdec/swfdec_as_interpret.c | 51 + test/trace/Makefile.am | 16 test/trace/chartoascii-4.swf |binary test/trace/chartoascii-4.swf.trace | 7 test/trace/chartoascii-5.swf |binary test/trace/chartoascii-5.swf.trace | 1011 ++++++++++++++++++++++++++++++++++++ test/trace/chartoascii-6.swf |binary test/trace/chartoascii-6.swf.trace | 1008

This might sound like an odd question.... but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then

Well guys here comes the fun part. I have a Microsoft access (VBA) application that interfaces with my SQL database. This app pulls of info from the SQL record and than picks up the phone and dials that locations number. I have purchased a few hundred NpaNxx's for my own use. I want get into too much detail there but no worries this is legal. I need to change my CID info on the fly. So I am

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

Hi, I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines. With only one of those, CID is shown. Beside that, everything is working OK. Lines have different providers and/or locations. All are located in France and CID Detection Method is ETSI FSK / Bell 202. If I'm connecting a TDM400-enabled Asterisk system, to one of those 2 non-working lines : it does work. The only

Hi, I`m having a problem I cannot explain. When dialing 555-555-5555 (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-5555, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine. Where could be the difference? Both are using the same context to dial out. Mike --------------

Hi list, My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get this error message on the asterisk console: == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' [Jul 8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable A long time ago my