similar to: Res: Asterisk with Nextone using H323

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone

Hi All, Thank's for your hint Tim Panton I could connect my asterisk machine to my oracle machine. I used unixODBC-2.2.11.tar.gz, oracle-instantclient-basic-10.2.0.3-1.i386.rpm, oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my asterisk machine. I can connect to my oracle machine with isql and in

Hi all, does anybody know any way that when it run "reload app_queue" in the asterisk cli it don't lose the informations from "show queue (queue name)" ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command "show queue (queue name)" show me

Hi People, I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome.... Thank`s all Everton Goularth Uberlandia - MG - Brazil _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as

Hi people, I'm tring to configure fax on my asterisk server. I'm using the instructions of: http://www.asteriskguru.com/tutorials/spandsp.html and the files app_rxfax.c, app_txfax.c and apps_Makefile_patch from: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ I have already configure this on other server that the operation system is CentOS 4 and all

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil _______________________________________________________

Hello All, How can I do to record my asterisk's CDR in a Oracle database? I have to use unixODBC? Can anybody send me a step to step to do this configuration? Thank's All Everton Goularth _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi there I hope someone can help - I am having a big problem getting calls cleared from several asterisk systems when RTP timeouts occur. It appears that asterisk doesn't send a BYE when it decides to terminate a call because of a RTP Timeout - is this a configuration problem? if so what need changing? or is this a bug/feature? if so is there a work around? The setup here is calls come from

Hi All, I need to limit outgoing calls in my sip peers... I tried to use "call-limit=1" in these peers in the sip.conf, but it didn't work... Here is my peer configuration in the sip.conf: [sip.broadvoice.com] accountcode=broadvoice type=peer dynamic=yes username=MYUSERNAME fromuser=MYUSERNAME authname=MYUSERNAME user=MYUSERNAME secret=xxxxxxxx host=sip.broadvoice.com

Hello All, Got one customer and he is getting disconnection within 15 seconds when he tries to make outbound calls. Initially, it was working fine without any glitches... Other customers on the same system are working fine, its just with this customer only. This is the error message thrown by Asterisk on the CLI: - Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum retries

Hi, I`m trying to use the agent snmp buit in the asterisk-1.4, but I can`t do this... I used this link to do it: http://www.voipphreak.ca/archives/382 but I can't... Does somebody know how to do this or knows a how-to to do this?? Thank's _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hello! I see many of you experience troubles with H323 stack. I am focusing on building H323-SIP Asterisk based softswitch with all codecs supported (including G729 and G723). I can setup Asterisk from scratch with H323 support or solve your h323 nightmare with existing asterisk system for for 100$. Contact me pls offline.

On 03/11/2019 15:06, Liste via samba wrote: > Am 03.11.2019 um 09:42 schrieb Rowland penny via samba <samba at lists.samba.org>: >> ?On 02/11/2019 23:18, Hilberg via samba wrote: >>> Hi >>> >>> The server suddenly changed the uid + gid. this happened to times, yesterday and the week after. The default group at example >>> The samba is a AD member

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.