similar to: Need some help with G729 passthru

I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many "answers." I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway,

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE!

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind

Hi, this is the second time that i post this, may be a wasnt clear the first time. Im having problems with an incoming peer after i upgraded asterisk from 1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this: register => @prepago-in [prepago-in] type=friend host=192.168.10.102 ; this is the cisco's ip context = from-external dtmfmode=rfc2833 insecure=very ; required for

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.

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi All, i have 3 asterisk server in a cluster using a cluster of mysql server via realtime, users can register via DNS SRV. I send/receive calls to an AS5400 via a SIP trunk defined on the realtime sip table, the trunk has call-limit=5. Problem i encountered is each of the 3 asterisk servers will 5 channels each to them instead of 5 for all 3 servers. Is there any solution to this?

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

Does anyone have any hardware recommendations for a device to take an incoming DS3 circuit and give me SIP that I can point to my Asterisk servers. Currently doing DS3 to Adtran but I want to get away from having PRI cards in all my Asterisk boxes. From looking around I've found some people using: Lucent Max TNT Dialogic IMG 1010 Cisco (Not sure which model would be best for this, the