similar to: Asterisk & Speechphone/Mandi

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip

Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070825/50bd7d63/attachment.htm

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I would like to send Multimedia Messaging (MMS) email (gateway) to my cell phone and have the from and subject be the callerid/calleridnam information from the voice mail message. I know there is a way to call another perl script or program up when an email message is written, but I am not a programmer. I know there could be a perl script or program that could run every minute and check the

Hi, I'm trying to use Hylafax without a modem. Is it possible to use t38modem to make Hylafax send and receive fax via Asterisk? If yes, how? I'm searching on internet but still haven't found anything useful. TIA GIorgio Incantalupo

Hi guys, I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs. If I receive i do not have any problem, but i'm not able to send put any fax, i get always the same error: txfax_exec: transmission done with ast_read(chan) == NULL Anyone has txfax working with asterisk 1.4? I try to download app_rxfax.c and app_txfax.c with the asterisk.patch file but without success.

Does realtime work reliably on Asterisk 1.2.24? Are there any definitive guides, I can only find bits and pieces here and there. Any accurate howtos would be of great help. I am missing func_realtime.so. Where does this file come from? Asterisk or asterisk-addons? I saw in one of the howtos that it is needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch lines in the

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

Hi, I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to show which devices are busy/not connected. The same phone worked with Asterisk 1.2.9.1. I would appreciate anyone who knows how to setup Asterisk 1.4.1 to behave as 1.2.9.1. TIA Giorgio Incantalupo

hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two

All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's & ATA's) via G729 -->Cisco Gateway-->Asterisk via G711. Any ideas? Has anybody done such an implementation or know

Hi all, A long time ago I built an Asterisk system that plays IceCast streams via moh. extensions.conf: Exten => moh,1,Set(SIP_CODEC=ulaw) Exten => moh,2,Answer Exten => moh,3,MusicONHold(test_new) Exten => moh,4,Hangup musiconhold.conf ; test_new [test_new] mode=custom application=/etc/mystreams/test_new.sh test_new.sh #!/bin/bash wget -q -T 120 -O -

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I'm still a newbie in Asterisk, just yesterday installed it for home use (so I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card. Anyway, I find it very hard to locate supporting information for Asterisk. User's Handbook is still a draft, this mailing list is not easily searchable, and any info out on Usenet is scarce. I spent some 30mins just trying to find out naming

Mandi a te! Does it work just with the CNAME? My config actually worked in the end, but I think after I added the spn entries. I was testing on a machine which didn't reflect the change, but probably because of caching (ifconfig /flushdns didn't help) ----- Original Message ----- > From: "samba" <samba at lists.samba.org> > To: "samba" <samba at

Hello everybody, i got and asterisk and a CCM configured thru SIP, and in the sip show peers appear Name/username Host Dyn Nat ACL Mask Port Status CCM 10.60.27.138 255.255.255.255 5060 OK (1 ms) but when i enabled sip debug in the CLI got this ------------------------------------------------------------ Sip read: SIP/2.0 400 Bad

Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA.

Hi, Is it possible to host call waiting service on Asterisk for a SIP device? What i am trying to achieve is that while a SIP user is busy on a call and a new call for that user comes in, asterisk should play the call waiting tone to that user. I have a vague idea that if i can get hold of the existing bridged channel when a subsequent call is received, i can then redirect that channel to