similar to: G729a and G729 interoperability

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

Hello, I'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also noticed that on

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the "old" binary) But there're a few questions: - should not the codec listed in the codec list when i enter "show codecs" ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar

Hello! I am having an odd problem in that when the caller dials extension "2" in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

Hello! The T.38 Digium Fax Driver sometimes responds with a successful sending of a fax, when in fact, the fax did not go through. 1. Where does this problem lie? 2. How to go about fixing it. Thanks, Elliot

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hello! I have a few sip devices and it is necessary for me to disable call-waiting and immediately return a busy signal if the sip's channel is busy on them. What is the procedure to do so in Asterisk 1.4? Thank you, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I have a 64 bit Asterisk system and am wondering how to use Digium's 32 bit fax driver. Is there some kind of emulation that can be used? Thanks! Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090604/b7f1d6c4/attachment.htm

Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is " john at phone.com" needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

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

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not