similar to: Happy 2007!!!

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Would like wish to ALL a Marry Christmas and a happy new year, full of peace, love, happinesses and much success. That let us have one excellent year of 2008. Best Regards Josue Conti.

Dear Ladies and Gentlemen how are you? I would like to wish everyone and all their families, may God continue to bless and always illuminating your steps, that this Christmas the joy of our Lord Jesus invade your home and participate in their family on this Christmas. May God bless you always! Merry Christmas and a Happy and Prosperous 2014. With Best Regards! Josue -------------- next part

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in "/dev/zap"(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : )

I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. -- Executing Page("SIP/2944093-5999", "SIP/3254107&SIP/3254105|") in new stack Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination '' supplied. -- Playing 'beep' (language

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Dear fellows, how are you? I?m offering TDM routes for Brazil (landline and mobile destinations) with low prices, TDM ccts (no GSM), ASR and ACD great. Pre paid, by paypal. If you have interest, please just let me know. With Best Regards Josue -------------- next part -------------- An HTML attachment was scrubbed... URL:

List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run

Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josu? -------------- next part -------------- An HTML attachment was

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josu?

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Hi, I am problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server it says It with C in iax and Server B it speaks with D in iax, but Server it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you authenticate sipspo

The hacker's lab will be organized similar to the one at the US 2016 meeting [0]. That means 1.5 hour sessions and theme tables in the different rooms. Each table (~10) will have a sign on it to represent a sub-project or area of LLVM. This will help newcomers and even active developers congregate together on specific topics. In addition to labeling tables, we are asking for volunteers to

Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get "Push message cannot be displayed" back from the Polycom phone, and all I am sending is the Polycom example : <PolycomIPPhone> <Data priority=?critical?> <h1> Fire Drill at 2pm </h1>

Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas.

Hi All Somebody knows as resolv the error below? Already I compiled asterisk-addons-1.2.3, but exactly thus it reports this error, could help me? -- Executing WaitMusicOnHold("SIP/3205-d9ef", "30") in new stack -- Started music on hold, class 'default', on SIP/3205-d9ef Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame

Hi all, I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. The only weird thing in the logs is this: May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its framing on the bus! May

Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling "E" and the TELCO says that asterisk would have to send signalling "F", as to make for asterisk to send signalling "F"? The TELCO says that the signalling "E" is