similar to: g729a compatibility

Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

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

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

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, Is SS7 and PRI in any way compatible in that if the interface is configured one it will work for the other (granted, it will not have any of the ISUP, etc. parameters available if the line is PRI) or are they two distince protocols that have incompatible signalling? Thanks, Elliot

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

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to

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

I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. Here is part of the content of sip.conf. [general] port = 5061 bindaddr = *.IP

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

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

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 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

Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as "tar" commands? Any advice will be appreciated tq

Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? -- wbr, Oleg

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="