sr-users February 2006

sr-users@lists.kamailio.org

189 participants
373 discussions

by Brijesh

Hi all I am new to OpenSER. I am using one of sip client. I want to connect sip clients through openSER proxy I have installed it and configured the clients but giving me error that user is not found Can anyone tell me what is reason for it ?

18 years, 9 months

[Serusers] Problem with ACK and BYE..

by pankaj.munjal＠wipro.com

Hi everyone, I've configured ser to work with semstalkflite module of sems. This is how it works: A user calls up sip:READOUTTHISTEXT@ser and ser should be able to play back "READ OUT THIS TEXT" using a media server(sems). Now it seems to be working fine, but, after having a INVITE-200OK-ACK dialog b/w ser and caller "A", ser is not absorbing the ACK. i,e, for some reason, Caller "A" keeps sending ACK again and again. FYI.. caller "A" uses X-lite. Here is the part of the ser config file that i've modified: if (method=="ACK"){ log(1,"Pankaj --> I've hit ACK!! "); # absorb ACKs break; }; Isn't it good enuf to absorb the incoming ACKs? Anyone with any pointers?? Thanks in advance, Pankaj Munjal The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

18 years, 9 months

[Serusers] firewall

by Toygun Mavinil

Hi, I use ser 0.9.6 on fc4 Ser Works with no problem but Firewall is closed I use ser with mediaproxy and sems (trying) Which ports i have to left open while enabling the firewall? Toygun

18 years, 9 months

Re: [Users] Using uac_replace_from to munge username between ser and Asterisk?

by Bogdan-Andrei Iancu

if you enable auto from_restore_mode, you do not need to perform any restore from script. Just replace the from in the initial INVITE and this is it - all replies and sequential request would be auto fixed (restore/replace). regards, bogdan Barry Flanagan wrote: > Bogdan-Andrei Iancu wrote: > >> Hi Barry, >> >> have you set auto from restoring? See: >> http://openser.org/docs/modules/1.1.x/uac.html#AEN75 > > > Yes, but I am not sure where it is supposed to go. > > I have the following in just before relaying to Asterisk: > > rewritehostport("XXX.XXX.XXX.XXX:5060"); > uac_replace_from("$fn","sip:$au_$ar@$fd"); > append_hf("P-hint: GATEWAY\r\n"); > t_relay("udp:XXX.XXX.XXX.XXX:5060"); > > > and I put in uac_restore_from(); just after the record_route() > > > with all the other modparams I have: > > modparam("uac","from_restore_mode","auto") > > > Thanks for the help. > > -Barry > > >> regards, >> bogdan >> >> Barry Flanagan wrote: >> >>> >>> So, the only way around it that I can see is to somehow have OpenSER >>> change the username to username_domain so that each will be unique. >>> >>> It looks like uac_from_replace should handle this. I have tried it, >>> and I can see that Asterisk does in fact get user_domain@domain in >>> the first invite, but thereafter for some reason OpenSER changes it >>> to just _@domain for subsequent requests. >>> >>> >>> Regards, >>> >>> -Barry >>> > >

18 years, 9 months

[Users] Receiving different VIA in INVITE and CANCEL

by Papadopoulos Georgios

Hello, I have an issue with some user agents, (Linksys PAP2 and Sipura 3000) sending different Via in CANCEL than in INVITE. Of course the result is that the caller hangs up and the callee keeps ringing. I tried to use modparam("tm", "via1_matching", 0) but it did not solve the problem. I have two questions: 1. Is the via1_matching in TM working as described or is it broken? 2. What can be causing this behavior of the user agent? Not all our users who have these user agents report this problem (actually very few have the problem). In our lab we tried same user agent, same firmware but could never reproduce this behavior no matter how bad we tried to mess up their configuration. Any help will be greatly appreciated. Thanks. George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

18 years, 9 months

[Serusers] No comunication between user through SER. S.O.S.

by Ernesto Garcia

Hi!!, I'm very new to this topic, I'm building a call center and I testing with Asterisk and SER, the problem is that when some registered user call to other user tha is registered too the phone of it don't detect the incoming call. Plase, what well be is the problem ? Ernesto

18 years, 9 months

Re: [Serusers] high-availibility setup using f5 bigip

by Jack Wei

I want to set up 2 SER boxes and 2 Asterisk boxes using 192.168.1.X IPs. I set up 2 VIPs, 1 for SER and 1 for Asterisk. SER redirect voice mail calls to Asterisk VIP. However, if you look at the SIP logs, Asterisk is using a 192.168.1.X IP for the contact address. Also, there is no audio since the SDP is using the 192.168.1.X IP. I've set the externip and localnet in Asterisk but nothing happened. My goal is to set up a load-balancing pools of SER and Asterisk behind a BIGIP switch. --- Charles Wang <lazy.charles(a)gmail.com> wrote: > I think that the range of this question is too large. > You should tell us what your scenario is. And tell us more about your > configurations. > > 2006/2/2, Jack Wei <cowlemon(a)yahoo.com>: > > hi, > > > > I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do > > load-balancing. I'm using Asterisk as a voicemail application only and > have > > successfully integrated SER with Asterisk without the switch. But when I > try > > to use the switch as a load-balancer, I get lots of NAT problems. Does > anyone > > know how to setup the switch and SER/Asterisk properly? > > > > Thanks, > > Jack > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > > Best Regards > Charles > Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 9 months

[Serusers] serctl, mysql and presence snapshot

by victoria beltran

>The reason is, that in this version (or some little older) has changed >the database structure and serctl is broken there (most serctl commands >are unusable). > >Vaclav Then, I believe the serweb will not work successful either with the presence snapshot, Is this right? Thus, neither the serweb nor the serctl tool works. Therefore, Are mysql commands the single way to manage (i.e create and drop users) the ser server?? thanks! --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

18 years, 9 months

[Serusers] serctl, mysql and presence snapshot

by victoria beltran

18 years, 9 months

[Serusers] Problems with DTMF using SER and Asterisk

by Anders Hansson

Hi all, I have a problem getting dtmf tones throgh to PSTN when using asterisk. The connections is SIP phone - Asterisk - SER - Cisco GW - PSTN When dialling PSTN numbers from the SIP phone and pressing the * or # button nothing is happening. I' ve set the dtmfmode=rfc2833 in asterisk. The problem is that to be able to control the asterisk voicemail the dtmfmode has to be =rfc2833, and the I can control the voicemail. But when calling external to PSTN the * and # button is not working. I do also have another problem, and that is when connecting a ATA box to the Asterisk. If I set dtmfmode=inband, voicemail in asterisk can be controlled but not the calls to external PSTN numbers. If I set dtmfmode=rfc2833, voicemail in asterisk do not work, but external calls to PSTN can be controlled with * and #. What can be the error? Please help, Thanks Anders

18 years, 9 months

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sr-users February 2006