similar to: Grandstreams can't call out with latest CVS

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

Hello list , I?d like to announce possible problems with migrating any version prior to 1.0.2 to 1.0.3. Pay attention : 1. Codecs Codecs names/description have been changed . For example : versions <= 1.0.2 voip*CLI> show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. 1 (1 << 0)

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

After browsing through bugs.digium.com I saw no mention of any work to get chan_sip or chan_sip2 to listen on tcp, as well as udp. Just curious is any-e-one working on such a patch at the moment?

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

The about-to-be-released Grandstream firmware now supports multiple ringtones, but (so far) I haven't been able to unearth any documentation as to how one uses them. Anyone out there know anything about this? I've googled, read the firmware READMEs and combed the GS site without any luck. Thx. B.

I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part

As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a

Just in case you guys haven't been paying attention Grandstream sliped in some diffrent colors on the IP phones and looks like they released the ATA-286 (Cisco is gonna have kittens I suspect) bkw

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing

I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks,

FYI. Haven't dug enough to be able to report any more, but re-fetched CVS to verify that sometime in the last few days CVS changes now break my GS phone. It appears to be at the RTP level. It seems to set the call up just fine, but no audio is passed back to the instrument. I reverted, and will try to play with this tomorrow unless someone else tells us it's fixed. Thx. B.