similar to: Signaling the status of the line on the phone

Displaying 20 results from an estimated 4000 matches similar to: "Signaling the status of the line on the phone"

2006 Mar 03
5
new beta Grandstream firmware HT488_496_386
http://grandstream.com/BETATEST/HT488_496_386/
2006 May 11
2
Paging and Auto Answer on Grandstream GXP2000
I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems
2006 Jan 31
3
Linking Asterisk Boxes with Sip
Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user
2006 Feb 14
3
Grandstream hold one way audio -URGENT
Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's
2006 May 08
3
Most comprehensive management?
I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or
2006 Jan 07
1
Some advice on routing DID's
Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote
2004 Apr 16
1
errors on Pri
I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! :-) Any ideas. Apr 16 10:40:12
2005 Oct 15
3
res_perl - Compiling error
Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"
2005 Oct 11
1
migrated to new ver on voip connection vs1 server voicemail problems
I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on
2005 Oct 13
1
call waiting not working on PAP2
Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Sep 29
2
Unable to send fax using BroadVoice
Has anyone had success sending faxes via a broadvoice byod account? Everything 'looks' to go as expected, but then my fax hangs up and I get a printout with Error 351. I am wondering if it is a codec issue or something. Any help will be great. Neri -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Sep 25
1
WRT54GP2 SIP server on LAN port
Hi, I'm trying to set up Asterisk behind my WRT54GP2 router that has a intergrated ATA box. My box are not locked in any way so I can access and change all settings. Now to the problem... I have gotten Asterisk to register with my provider and everything works just well.. Now it's time to get the intergrated ATA to connect to asterisk. But the asterisk box in located on the LAN ports of
2006 Jun 08
2
Phone recommendations?
Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel
2006 Jun 08
2
FreePBX 2.1.0: Manually rewriting extensions_additional.conf
Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the
2006 Mar 03
3
Sipura RMA
Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just
2005 Oct 03
4
R: Diva
Which models of Diva could work with CAPI and asterisk? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di gw@adcomcorp.com Inviato: sabato 1 ottobre 2005 23.46 A: asterisk-users@lists.digium.com Oggetto: RE: [Asterisk-Users] Diva Nope. At least I tried and never could get it
2006 Mar 30
1
caller anounce
I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the
2006 Mar 31
1
Play wav while in connection with a caller
Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints
2006 May 09
1
A@H Memory Limits
Hi I have the latest a@h installed and everything is working perfectly. I have been told by a collegue that a@h doesn't use the full potential of the machine it is installed on i.e. the CPU & Memory, unless the kernel has modified. He is unsure where he heard this from and I wonderd if anyone had any other information about this or knew where I could find some? Many Thanks Scott
2006 May 31
1
Upgrade ONLY asterisk from an AAH install
Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of