similar to: * and signaling (clarification)

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

> Thanks for the reply. > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. I > am not sure if OutboundProxy has to be configured to have it working fine. > Or this just happened to me? What is your ATA's software? > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > As per ATA, it is by default using rfc2833.

On Thursday, January 22, 2004 9:55 PM, Jess Magnaye [SMTP:jess@arretni.com] wrote: > in telco world, there's like 64kbps per channel and voice can be > carried on a 16kbps channel. is it possible to configure asterisk to > make 4 extensions (ATAs example), to call out using single FXO port > at the same time? if that is possible, then is it also possible to > make t1-pri to

I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this

You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens. -d > Message: 10 > From: "Bartosz Jozwiak" <bartek@cq-link.sr> >

Here's my sip debug output. anybody knows why Cisco sent * is CANCEL msg? Can someone tell me what ATA version are they using? Maybe this is also another issue.. I am using v2.16. This is using G711ulaw. Sip read: > SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e From: "Jess" <sip:6882332@mydomain.com>;tag=as6818ebfb To:

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone.

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

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

I have cisco phones running SCCP, and a cisco 2600 with FXO I'm using for PSTN access. I can dial out, but inbound calls are not ringing a phone. Please see my config In the 2600 I'm PLAR'ing the line and I have a SIP Dial-Peer for 4001 voice-port 1/1/0 output attenuation 0 echo-cancel coverage 32 no comfort-noise timing hookflash-out 50 connection plar opx 4001

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed