similar to: Caller ID on TelaSIP SIP Channel

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXXXXXX) ? It has always worked for the Teliax lines. BUT--- It doesn't have a problem

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

First, I am not a RedHat or linux newbie. I simply have not had to do what I am getting ready to do, and I want to see if I am going to run into a problems... My HDA drive is failing (I can hear the occasional click from it and I am seeing Smart errors, the transfer rate is slow but all my data is there). I have 3 partitions on it, the /, /boot/ and my game servers (this is also the drive the

I've had the same excellent responsiveness from telasip, on the rare occasion that I've had issues. YMMV -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Vile Sent: Thursday, April 06, 2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telasip

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

I have been having a problem with the test releases...I am now using Test 59. When I delete a message using SquirrelMail (IMAP) the message stays in the message list....but you can't open it anymore because it can't be found, if I logout and log back in...the message is gone from the list. I thought it could be a problem with the cache, so I added mail_never_cache_fields = MessagePart

My DID with Telasip is disconnected and my Asterisk box won't register with them. Anyone else having problems with them? Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050814/886ceb38/attachment.htm

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

Does anyone have a patch or information resource on implementing POP before SMTP with sendmail and DoveCot? I implemented it using a qpopper patch before we swiched to DoveCot for Maildirs. We have been using DoveCot for almost a year now and love it better than any other IMAP or POP3 server. Thanks for the info, Doug Eubanks SIMflex Internet Support support at simflex.com SIMflex Telephone

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/7e1e68d1/attachment.htm

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as