similar to: SIP wierdness after upgrade from 0.7.1 to CVS

Displaying 20 results from an estimated 1200 matches similar to: "SIP wierdness after upgrade from 0.7.1 to CVS"

2004 Jan 30
1
Words for Allison(?)
I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wish). Any chance of getting these? While I'm on the subject, I'd be very interested in a
2004 Aug 13
3
Will this ISDN card work for me?
I'm looking for an ISDN card that will work for me using either the i4l or capi with asterisk under Linux. I'm in the US, so I need an ISDN-U interface. Can anyone tell me if this card will work for me? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=14922&item=6700879965&rd=1&ssPageName=WDVW There are a lot available, and for what seems to be a good price, if
2004 Jul 13
0
Looking for US ISDN card...
Not having a whole lot of luck... I've decided I need to open up the search to cards with S/T interfaces and just find an NT1, too. Can someone with experience give me some pointers what would be easy to find, and easy to configure under Linux? I've stumbled across an Eicon card that's just labeled "DIVA T/A PCI", both on stickers and printed on the circuit board. Is this
2004 Aug 19
1
SpanDSP/RxFax help...
I've seen people mention that they have fax reception working with Asterisk, spandsp, and app_rxfax. I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. The fax part of the dialplan is pretty
2004 Jun 01
5
Some (lack of) answers regarding the wakeup call application...
Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not perfectly happy with, though. There are two AGI scripts, written in Perl, which handle (a) scheduling, confirming, and cancelling a wakeup call, and (b) the
2009 Dec 23
1
Problems with chan_sip
Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of
2004 May 06
0
Unable to find the source of the error: bad file descriptora
Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.
2010 Jun 10
1
warning : sip_xmit
I'm getting a lot of these on the CLI : [Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:38] WARNING[4286]:
2011 Mar 15
2
Some errors
Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and
2004 Aug 11
0
Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted
When I try to make a call using the Mediatrix 1204 is showed on the CLI: -- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack -- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new stack Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x81 40c5c (len 794) to 192.168.199.5 returned -1: Operation not
2004 Sep 16
1
Unable to dial using SIP using FWD and iConnectHere
Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for
2010 Jun 25
1
sip_xmit: sip_xmit returned -1: Operation not permitted
Hello, my Asterisk CLI is flooded with the following message : [Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:05]
2004 Jan 08
0
Error messages during Registeration on CVS Version CVS-01/08/04-14:20:34
An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/748d21b3/attachment.htm -------------- next part -------------- Hello I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34) As a result IP Phone don't register with the Asterisk. Is it broken ? How can I
2011 Apr 04
2
WARNING chan_sip.c:3115 __sip_xmit
Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",
2004 Dec 09
0
New batch of phrases from Allison
See the following: http://bugs.digium.com/bug_view_page.php?bug_id=0003006 I'll be collecting suggestions and requests, as well as donations, to pay Allison to do another batch of new phrases for Asterisk. The result will be donated to the Asterisk community, and submitted to the powers that be for inclusion in CVS. Please email me or add a comment on the bug tracker with your suggestions.
2011 Feb 18
2
no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
2005 Jul 18
0
chan_sip.c:939 __sip_xmit warning
Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address
2011 Jun 07
1
tls/srtp: sip_xmit error: returned -2
I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:
2004 Sep 13
0
Registering asterisk with FWD
Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922
2004 Dec 02
0
Newby with no idea
Hi folks, thanks for your help with my last question re: japanese FXO. It doesnt sound very compatible so I will use a SIP FXO gateway then. Untill I find one, im just trying to get my 2 cisco SIP phones talking to my * server. just as a learning experience for now. heres what I have so far: 2 Cisco 7960's both using DHCP and both registering with my SIP proxy server (Brekeke OnDo on