search for: comellas

...far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas (juanjo@comellas.com.ar)

.....o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is workin...

...n=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c2 1.15.3-4 -- Juan Jose Comellas (juanjo@comellas.com.ar)

...to active calls after the dialplan configuration is updated? - Can we do partial updates of the dialplan (e.g. update a specific context instead of the whole dialplan configuration)? - Can Asterisk have its dialplan in a database instead of having it always in memory? Thanks. -- Juan Jose Comellas (juanjo@comellas.com.ar)

...7) signalling=fxs_ks language=en callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 group=1 useincomingcalleridonzaptransfer=yes callerid=asreceived context=pstn-inbound-voice channel => 2-7 Thanks -- Juan Jose Comellas (juanjo@comellas.com.ar)

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

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?

...apparently they only allow two channels. Is there any special precaution that I have to be aware of when doing this? Do I have to masquerade the channels that are inserted into the conference? The channels will mainly use SIP (maybe IAX2 too occasionally). Thanks for your help. -- Juan Jose Comellas (juanjo@comellas.com.ar)

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Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance.

Hello - I''m interested in leveraging wxRuby on my PowerPC-based iBook. I''ve got Mac OS 10.3.9 and I installed wxRuby 1.9.2 using gems. However, when I try to run samples such as minimal.rb, I get the following error: dyld: ruby Undefined symbols: _DataBrowserChangeAttributes _PMGetDuplex _PMSetDuplex _TXNGetCommandEventSupport _TXNSetCommandEventSupport

...ffix or operands invalid for `popf' /tmp/ccXxGHg6.s:15: Error: suffix or operands invalid for `pushf' /tmp/ccXxGHg6.s:16: Error: suffix or operands invalid for `pop' /tmp/ccXxGHg6.s:17: Error: suffix or operands invalid for `popf' make[2]: *** [testcpuid.lo] Error 1 -- Juan Jose Comellas (juanjo@comellas.com.ar)

Hi all, I was reading http://bugs.digium.com/view.php?id=2639 and it seems that anthm's great native MoH patch only works on HEAD. Does anyone have a version of the native MoH patch that works on 1.0.8? If so please point me to its location or email it off-list. Thanks and regards, Patrick

Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

...tinc v1.0.3 does have a bug that prevents broadcast packets to be sent when the switch mode is used. I actually found that while trying to set an vpn using the Debian unstable packages of tinc (which are v1.0.3). Are any plans to fix this in an official release soon? Regards -- Eduardo D?az Comellas ediaz@ultreia.es

I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The

...; > > Today's Topics: > > 1. Asterisk ended with exit status 1 (Federico > Alves) > 2. Re: Re: teliax [Was: LiveVoip is Bankrupt] > (Rich Adamson) > 3. RE: Polycom & VPN trouble (gw@adcomcorp.com) > 4. Re: Native MoH patch for 1.0.8? (Juan Jose > Comellas) > 5. Re: Polycom & VPN trouble (Tim Pushor) > 6. Re: Re: teliax [Was: LiveVoip is Bankrupt] > (r00t) > 7. Newbie Confusion on Call Forward and > DBput/DBdel (Jeffrey Starin) > 8. Eicon equipment, BRI Server or PRI? > (gw@adcomcorp.com) > 9. Re: Level 3 SI...

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

Last week I migrated some of our servers to Asterisk 1.4.18 and we started seeing audio drops of several seconds during SIP calls. After investigating it we noticed that Asterisk was increasing the RTP timestamps abnormally during a conversation. I'm including a text file with a subset of the data collected by Wireshark that shows the problem (I have the complete packet capture if anybody