similar to: any SIP client for BlackBerry?

client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo < rodolfo.alcazar at padep.org.bo> wrote: > Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. > Could be what you want. > > Greets! > > Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: > > TianLun, > > > > I

Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only "333" What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha

Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP->PSTN

I have currently following the instructions at- http://www.voip-info.org/wiki/view/QoS+Cisco I am using a Cisco 877 router with an ADSL2+ connection. My upstream bandwidth is around 1Mbps, and my downstream around 4-5Mbps. I have two PSTN side interfaces - atm0 and dialer0 (dialer0 is where the ISP details are configured and the interface that the WAN side IP is applied on) Now which

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hi. Does anyone knows how to trigger a phone call from a bash command? Thx! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha Contra La Pobreza - PADEP Av. S?nchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email:

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --

I'm using it rather successfully. Not perfect, but it works. It is limited to WiFi connectivity... at least here in Spain I cant get either client to work over 3G. I'm using Fring and Truphone. Although I have only configured a SIP to my Asterisk with Fring. Skype works fine. We tested with several Nokia 5800 (EM) using Fring. Call quality is worse. At best, we have a 1+ second delay.

Hi, Is anyone using Fring as a SIP client to an Asterisk server ? A prospective customer of mine is asking to integrate its iphones with an Asterisk server and after googling, I still have some unanswered questions : 1. Which codecs are available when calling from fring ? 2. Is it easy and natural to change your presence status (available, busy, ...) with Fring or will users prefer to use

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi, I've been getting this bug in 1.0alpha4. It happens every time the user checks his Inbox. I'm using mbox btw. Can I get any hint on why this is happening? Dec 9 09:08:39 lasige-serv dovecot: imap(pjsousa): file mail-index-sync-ext.c: line 155 (get_ext_header): assertion failed: (memcmp((char *)(ext_hdr + 1), ext->name, strlen(ext->name)) == 0) Dec 9 09:08:39 lasige-serv

Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

Hi, I have installed a CCNP test application that came with a CCNP book but the actual text of the questions is missing. The wine version is 1.2.2, the OS is xubuntu In the quiz preferences the two font options are arial and times new roman I am guessing that it just can't find these fonts, but i am not exactly sure where i should be putting them. I have installed MStruetype fonts but

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Hi, I just want to deny FOP originated calls in TRIXBOX. All remaining operations (hanging up, transferring, etc) should go. Where is that option in TRIXBOX (already googled, nothing, checked config files but cant find that option)? Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Hi,

Hey guys, I configured Samba do be the domain controller for my network and to share folders. the folder sharing works great. The problem is that the domain function does not work at all. I cannot join the domain from any workstation. It just says that the controller cannot be contacted. I ran an Ethereal sniff on the packets and the computer that i want to be PDC is sending ICMP Destination

Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the