similar to: Inbound Calls Codec

I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only

On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote: >Gary wrote: > >>I have noticed lack of codec negotiation with calls thru a registrated >>asterisk box. >> >>No seen problems with outbound calls, (though I haven't specifically >>tried it), but the problem exists inbound. >> >>Easiest method for testing this was ring in via a sip client set

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

Hi, I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the "primary" Asterisk server where all the SIP phone extensions are registered. The IAX trunk settings are something like this (all servers have this identical except for the "host" field): [inbound] deny=all allow=alaw allow=gsm type=friend

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I made this change in my sip.conf file, I removed allow=gsm, allow=alaw and now everthing works great. Chris Tuska [general] disallow=all allow=gsm allow=ulaw allow=alaw ; My PSTN Service provider [Sipmedia] disallow=all allow=gsm allow=ulaw allow=alaw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I have Cisco 7960's, Cisco 2950 Switch. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone is disconnects at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone seen this? Thanks for the help,

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to

On Fri, 11 May 2018, Joshua Colp wrote: >> In the above example, even though the INVITE/SDP says they prefer gsm >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose >> to use gsm or ulaw? > > Yes. > >> Can it be asymmetrical? They send gsm and I send ulaw? > > Technically, yes. In practice it's a bit iffy - specifically because

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm

Hi all, I would be most grateful if someone would review my plans for my new phone system and comment on areas of expected trouble and advice on what to do better. Instead of moving our Panasonic KX-TD1232/TVS200 system (ugh...) to our new location, we've decided to jump into IP telephony with *. But we are new to * (but not Linux), so we're trying to learn as much as we can before we

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw

Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example

Need some help understanding codec preferences: I have 2 asterisk servers. Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and allow=ulaw in iax.conf Server 2 receives calls and routes them to server 1. It has the same allow lines. We receive calls from a phone co and route them via server 2 to server 1. The calls originate in g729 and everything works fine. Now I want to take