similar to: chan_sip not 100% RFC3665 compliant - re-REGISTERs fail.

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Hi all, Today I've stumbled upon a very strange behaviour with an analog fxs/fxo gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html) connected to a CVS HEAD(from today) Asterisk server. This manifested itself after enabling the CallerID on the pstn lines connected to the FXO ports of the module. Both FXO modules have their own sip username/passwords and are registered to the

Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any

Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can load addpac. Is there anyway that can I upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the image: # -------------- next part -------------- An

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

I am trying to connect an ADDPAC 200 to * using SIP protocol. Problem is the documentation for the ADDPAC is not very helpful (they don't mention SIP at all, seems to be a newer firmware). I think I got the * part working but the ADDPAC setup is the problem, I will appreciate if somebody can post a working example or point me to documentation in english (found stuff in russian and korean so

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

What is the proper way of adding hints to multiple extensions? In my case extensions are the same as the sip usernames, while as per http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence exten => 1234,hint,SIP/1234 works, exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use ${EXTEN} here... Any hints? Vahan -------------- next part -------------- A

http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf

This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there:

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

Hi, I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64? I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection. Any ideas? Vahan

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,