similar to: Cisco 7960 x g729 x Unable to create/find channel

Hi, I have installed chan_sip on asterisk-1.0.3 / 5 (tried both, same result). My sip phone registers fine. But when dialing a number, I get: Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to create/find channel ... Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 384534305@192.168.1.20 for seqno 219 (Non-critical Response)

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.

Hello All, I always used a compiled version for a x86 system >From http://kvin.lv/pub/Linux/Asterisk/ , and Works fine. At this time , I'm using a Pentium IV Dual core, Running Centos 4.3. I tried to install the 64 bits Compiled version but has a translation time > 20ms. Is it correct ? I also tried to make my self compilation and apply The patch but the patch doesn't work on ICC

I had everything working fine until today. Today the Sipura cannot dial anywhere. I just get the following: Feb 10 12:48:18 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:19 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:35 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

Hi, I have configured the cdr throught ODBC with this files: /etc/cdr_odbc.conf [global] dsn=asterisk2 ;loguniqueid=yes dispositionstring=yes table=cdr ;"cdr" is default table name usegmtime=no ; set to "yes" to log in GMT hrtime=yes ;Enables microsecond accuracy with the billsec and duration fields /etc/cdr.conf [general] enable=yes unanswered =

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

Hi, I have made a fresh install of asterisk-1.6.2.10 and when I register my soft phone it gives following error. Rest are default configurations. 32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER

All of a sudden, on three different systems, the "server_ip_address/sid_name" syntax in database.yml has stopped working for me. I can''t even do a OCI8.new(blah blah blah) statement from an IRB shell. I get "ORA-12541: TNS:no listener", or some variant, depending on how I phrase it. On systems where I have a real deal client installed, and OCI can find the tnsnames

Hello everybody. I''m running linux 2.6.19 with nth match to alternatively snat outgoing connections to two different ip addresses for load balancing between two adsl lines: Here is: $IPTABLES -t nat -A POSTROUTING -s my_ip --protocol tcp -m multiport --dports 80,443 -m statistic --mode nth --every 2 -j SNAT --to adslA $IPTABLES -t nat -A POSTROUTING -s my_ip --protocol tcp -m multiport

Hi, I''m looking to combine a couple of fact names with the value of a class parameter to create and lookup the resulting fact''s value. Is that possible? For example, my class will take the parameter "my_default_nic" from beyond. So I know that as long as $my_default_nic exists on the client, then so will facts like macaddress_<NIC>, netmask_<NIC>,

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Hello, I''m trying to install the upload_progress gem and i can''t see the upload progress status. My config is : apache2 with mod_proxy mongrel rails upload.rb : ############## require ''rubygems'' require ''drb'' require ''gem_plugin'' GemPlugin::Manager.instance.load ''mongrel'' => GemPlugin::INCLUDE

I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when