similar to: Snom 300 MWI

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

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox=1234@default and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. Any idea on how to get this to behave

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI

I have a Sipura 962 endpoint on Asterisk 1.4 where the MWI works perfectly, however my theory is that it should be broken. Obviously I'm wrong but "Sip show subscriptions" does not show the endpoint subscribing to the MWI status on Asterisk, even though all of the other endpoints on the system DO subscribe for their respective mailboxes, including SNOM & Polycom endpoints.

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706] type=friend host=xxx.xxx.xxx.xxx (IP of proxy) port=5060 nat=no mailbox=1234 at customer subscribemwi=no

Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hi, How have the same sip username in several realms ? For now, I must add the realm prefix in the sip username of chan_sip. For example: [lg_2540] amaflags = default call-limit = 10 host = dynamic language = en_US context = lg_default callerid = "LG" <2540> secret = XXXXXXXXXXXXXXXXXXXXXXXXXX type = friend subscribemwi = no mohsuggest = default qualify = yes

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100714/d268d1f8/attachment.htm

We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. While not exhaustive, these are the ATAs that don't work: Linksys SPA2102 Linksys PAP2T-3.1.15 Thomson 780 Thomson 784 Unfortunately, this

Hello list, what is the difference between mohsuggest & mohinterpret when defining a SIP peer ?! If a certain SIP peer puts another channel on hold, what field then determines the moh class that Asterisk will choose to play to that channel ? If I take the test and call from peer A to peer B, and peer A puts peer B in hold, then the class of peer B is taken... that's not what I want.

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

Hi list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already "solved" (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes